diff options
Diffstat (limited to '')
125 files changed, 10467 insertions, 11898 deletions
diff --git a/src/decoder/AdPlugDecoderPlugin.cxx b/src/decoder/AdPlugDecoderPlugin.cxx new file mode 100644 index 000000000..5c04e116d --- /dev/null +++ b/src/decoder/AdPlugDecoderPlugin.cxx @@ -0,0 +1,144 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "AdPlugDecoderPlugin.h" +#include "tag/TagHandler.hxx" +#include "DecoderAPI.hxx" +#include "CheckAudioFormat.hxx" +#include "util/Error.hxx" + +#include <adplug/adplug.h> +#include <adplug/emuopl.h> + +#include <glib.h> + +#include <assert.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "adplug" + +static unsigned sample_rate; + +static bool +adplug_init(const config_param ¶m) +{ + Error error; + + sample_rate = param.GetBlockValue("sample_rate", 48000u); + if (!audio_check_sample_rate(sample_rate, error)) { + g_warning("%s", error.GetMessage()); + return false; + } + + return true; +} + +static void +adplug_file_decode(struct decoder *decoder, const char *path_fs) +{ + CEmuopl opl(sample_rate, true, true); + opl.init(); + + CPlayer *player = CAdPlug::factory(path_fs, &opl); + if (player == nullptr) + return; + + const AudioFormat audio_format(sample_rate, SampleFormat::S16, 2); + assert(audio_format.IsValid()); + + decoder_initialized(decoder, audio_format, false, + player->songlength() / 1000.); + + int16_t buffer[2048]; + const unsigned frames_per_buffer = G_N_ELEMENTS(buffer) / 2; + DecoderCommand cmd; + + do { + if (!player->update()) + break; + + opl.update(buffer, frames_per_buffer); + cmd = decoder_data(decoder, NULL, + buffer, sizeof(buffer), + 0); + } while (cmd == DecoderCommand::NONE); + + delete player; +} + +static void +adplug_scan_tag(enum tag_type type, const std::string &value, + const struct tag_handler *handler, void *handler_ctx) +{ + if (!value.empty()) + tag_handler_invoke_tag(handler, handler_ctx, + type, value.c_str()); +} + +static bool +adplug_scan_file(const char *path_fs, + const struct tag_handler *handler, void *handler_ctx) +{ + CEmuopl opl(sample_rate, true, true); + opl.init(); + + CPlayer *player = CAdPlug::factory(path_fs, &opl); + if (player == nullptr) + return false; + + tag_handler_invoke_duration(handler, handler_ctx, + player->songlength() / 1000); + + if (handler->tag != nullptr) { + adplug_scan_tag(TAG_TITLE, player->gettitle(), + handler, handler_ctx); + adplug_scan_tag(TAG_ARTIST, player->getauthor(), + handler, handler_ctx); + adplug_scan_tag(TAG_COMMENT, player->getdesc(), + handler, handler_ctx); + } + + delete player; + return true; +} + +static const char *const adplug_suffixes[] = { + "amd", + "d00", + "hsc", + "laa", + "rad", + "raw", + "sa2", + nullptr +}; + +const struct decoder_plugin adplug_decoder_plugin = { + "adplug", + adplug_init, + nullptr, + nullptr, + adplug_file_decode, + adplug_scan_file, + nullptr, + nullptr, + adplug_suffixes, + nullptr, +}; diff --git a/src/decoder/AdPlugDecoderPlugin.h b/src/decoder/AdPlugDecoderPlugin.h new file mode 100644 index 000000000..9fdf438aa --- /dev/null +++ b/src/decoder/AdPlugDecoderPlugin.h @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_ADPLUG_H +#define MPD_DECODER_ADPLUG_H + +extern const struct decoder_plugin adplug_decoder_plugin; + +#endif diff --git a/src/decoder/AudiofileDecoderPlugin.cxx b/src/decoder/AudiofileDecoderPlugin.cxx new file mode 100644 index 000000000..1ee57de4a --- /dev/null +++ b/src/decoder/AudiofileDecoderPlugin.cxx @@ -0,0 +1,265 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "AudiofileDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "util/Error.hxx" + +#include <audiofile.h> +#include <af_vfs.h> +#include <assert.h> +#include <glib.h> +#include <stdio.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "audiofile" + +/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */ +#define CHUNK_SIZE 1020 + +static int audiofile_get_duration(const char *file) +{ + int total_time; + AFfilehandle af_fp = afOpenFile(file, "r", nullptr); + if (af_fp == AF_NULL_FILEHANDLE) { + return -1; + } + total_time = (int) + ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK) + / afGetRate(af_fp, AF_DEFAULT_TRACK)); + afCloseFile(af_fp); + return total_time; +} + +static ssize_t +audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length) +{ + struct input_stream *is = (struct input_stream *) vfile->closure; + + Error error; + size_t nbytes = is->LockRead(data, length, error); + if (nbytes == 0 && error.IsDefined()) { + g_warning("%s", error.GetMessage()); + return -1; + } + + return nbytes; +} + +static AFfileoffset +audiofile_file_length(AFvirtualfile *vfile) +{ + struct input_stream *is = (struct input_stream *) vfile->closure; + return is->GetSize(); +} + +static AFfileoffset +audiofile_file_tell(AFvirtualfile *vfile) +{ + struct input_stream *is = (struct input_stream *) vfile->closure; + return is->GetOffset(); +} + +static void +audiofile_file_destroy(AFvirtualfile *vfile) +{ + assert(vfile->closure != nullptr); + + vfile->closure = nullptr; +} + +static AFfileoffset +audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset offset, int is_relative) +{ + struct input_stream *is = (struct input_stream *) vfile->closure; + int whence = (is_relative ? SEEK_CUR : SEEK_SET); + + Error error; + if (is->LockSeek(offset, whence, error)) { + return is->GetOffset(); + } else { + return -1; + } +} + +static AFvirtualfile * +setup_virtual_fops(struct input_stream *stream) +{ + AFvirtualfile *vf = new AFvirtualfile(); + vf->closure = stream; + vf->write = nullptr; + vf->read = audiofile_file_read; + vf->length = audiofile_file_length; + vf->destroy = audiofile_file_destroy; + vf->seek = audiofile_file_seek; + vf->tell = audiofile_file_tell; + return vf; +} + +static SampleFormat +audiofile_bits_to_sample_format(int bits) +{ + switch (bits) { + case 8: + return SampleFormat::S8; + + case 16: + return SampleFormat::S16; + + case 24: + return SampleFormat::S24_P32; + + case 32: + return SampleFormat::S32; + } + + return SampleFormat::UNDEFINED; +} + +static SampleFormat +audiofile_setup_sample_format(AFfilehandle af_fp) +{ + int fs, bits; + + afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); + if (!audio_valid_sample_format(audiofile_bits_to_sample_format(bits))) { + g_debug("input file has %d bit samples, converting to 16", + bits); + bits = 16; + } + + afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, + AF_SAMPFMT_TWOSCOMP, bits); + afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); + + return audiofile_bits_to_sample_format(bits); +} + +static void +audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) +{ + AFvirtualfile *vf; + int fs, frame_count; + AFfilehandle af_fp; + AudioFormat audio_format; + float total_time; + uint16_t bit_rate; + int ret; + char chunk[CHUNK_SIZE]; + + if (!is->IsSeekable()) { + g_warning("not seekable"); + return; + } + + vf = setup_virtual_fops(is); + + af_fp = afOpenVirtualFile(vf, "r", nullptr); + if (af_fp == AF_NULL_FILEHANDLE) { + g_warning("failed to input stream\n"); + return; + } + + Error error; + if (!audio_format_init_checked(audio_format, + afGetRate(af_fp, AF_DEFAULT_TRACK), + audiofile_setup_sample_format(af_fp), + afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK), + error)) { + g_warning("%s", error.GetMessage()); + afCloseFile(af_fp); + return; + } + + frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); + + total_time = ((float)frame_count / (float)audio_format.sample_rate); + + bit_rate = (uint16_t)(is->GetSize() * 8.0 / total_time / 1000.0 + 0.5); + + fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); + + decoder_initialized(decoder, audio_format, true, total_time); + + DecoderCommand cmd; + do { + ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, + CHUNK_SIZE / fs); + if (ret <= 0) + break; + + cmd = decoder_data(decoder, nullptr, + chunk, ret * fs, + bit_rate); + + if (cmd == DecoderCommand::SEEK) { + AFframecount frame = decoder_seek_where(decoder) * + audio_format.sample_rate; + afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame); + + decoder_command_finished(decoder); + cmd = DecoderCommand::NONE; + } + } while (cmd == DecoderCommand::NONE); + + afCloseFile(af_fp); +} + +static bool +audiofile_scan_file(const char *file, + const struct tag_handler *handler, void *handler_ctx) +{ + int total_time = audiofile_get_duration(file); + + if (total_time < 0) { + g_debug("Failed to get total song time from: %s\n", + file); + return false; + } + + tag_handler_invoke_duration(handler, handler_ctx, total_time); + return true; +} + +static const char *const audiofile_suffixes[] = { + "wav", "au", "aiff", "aif", nullptr +}; + +static const char *const audiofile_mime_types[] = { + "audio/x-wav", + "audio/x-aiff", + nullptr +}; + +const struct decoder_plugin audiofile_decoder_plugin = { + "audiofile", + nullptr, + nullptr, + audiofile_stream_decode, + nullptr, + audiofile_scan_file, + nullptr, + nullptr, + audiofile_suffixes, + audiofile_mime_types, +}; diff --git a/src/decoder/AudiofileDecoderPlugin.hxx b/src/decoder/AudiofileDecoderPlugin.hxx new file mode 100644 index 000000000..59c09c006 --- /dev/null +++ b/src/decoder/AudiofileDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_AUDIOFILE_HXX +#define MPD_DECODER_AUDIOFILE_HXX + +extern const struct decoder_plugin audiofile_decoder_plugin; + +#endif diff --git a/src/decoder/DsdLib.cxx b/src/decoder/DsdLib.cxx new file mode 100644 index 000000000..7135c9903 --- /dev/null +++ b/src/decoder/DsdLib.cxx @@ -0,0 +1,171 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* \file + * + * This file contains functions used by the DSF and DSDIFF decoders. + * + */ + +#include "config.h" +#include "DsdLib.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "util/bit_reverse.h" +#include "tag/TagHandler.hxx" +#include "tag/TagId3.hxx" +#include "util/Error.hxx" + +#include <unistd.h> +#include <string.h> +#include <stdio.h> /* for SEEK_SET, SEEK_CUR */ + +#ifdef HAVE_ID3TAG +#include <id3tag.h> +#endif + +bool +dsdlib_id_equals(const struct dsdlib_id *id, const char *s) +{ + assert(id != nullptr); + assert(s != nullptr); + assert(strlen(s) == sizeof(id->value)); + + return memcmp(id->value, s, sizeof(id->value)) == 0; +} + +bool +dsdlib_read(struct decoder *decoder, struct input_stream *is, + void *data, size_t length) +{ + size_t nbytes = decoder_read(decoder, is, data, length); + return nbytes == length; +} + +/** + * Skip the #input_stream to the specified offset. + */ +bool +dsdlib_skip_to(struct decoder *decoder, struct input_stream *is, + goffset offset) +{ + if (is->IsSeekable()) + return is->Seek(offset, SEEK_SET, IgnoreError()); + + if (is->GetOffset() > offset) + return false; + + char buffer[8192]; + while (is->GetOffset() < offset) { + size_t length = sizeof(buffer); + if (offset - is->GetOffset() < (goffset)length) + length = offset - is->GetOffset(); + + size_t nbytes = decoder_read(decoder, is, buffer, length); + if (nbytes == 0) + return false; + } + + assert(is->GetOffset() == offset); + return true; +} + +/** + * Skip some bytes from the #input_stream. + */ +bool +dsdlib_skip(struct decoder *decoder, struct input_stream *is, + goffset delta) +{ + assert(delta >= 0); + + if (delta == 0) + return true; + + if (is->IsSeekable()) + return is->Seek(delta, SEEK_CUR, IgnoreError()); + + char buffer[8192]; + while (delta > 0) { + size_t length = sizeof(buffer); + if ((goffset)length > delta) + length = delta; + + size_t nbytes = decoder_read(decoder, is, buffer, length); + if (nbytes == 0) + return false; + + delta -= nbytes; + } + + return true; +} + +/** + * Add tags from ID3 tag. All tags commonly found in the ID3 tags of + * DSF and DSDIFF files are imported + */ + +#ifdef HAVE_ID3TAG +void +dsdlib_tag_id3(struct input_stream *is, + const struct tag_handler *handler, + void *handler_ctx, goffset tagoffset) +{ + assert(tagoffset >= 0); + + if (tagoffset == 0) + return; + + if (!dsdlib_skip_to(nullptr, is, tagoffset)) + return; + + struct id3_tag *id3_tag = nullptr; + id3_length_t count; + + /* Prevent broken files causing problems */ + const goffset size = is->GetSize(); + const goffset offset = is->GetOffset(); + if (offset >= size) + return; + + count = size - offset; + + /* Check and limit id3 tag size to prevent a stack overflow */ + if (count == 0 || count > 4096) + return; + + id3_byte_t dsdid3[count]; + id3_byte_t *dsdid3data; + dsdid3data = dsdid3; + + if (!dsdlib_read(nullptr, is, dsdid3data, count)) + return; + + id3_tag = id3_tag_parse(dsdid3data, count); + if (id3_tag == nullptr) + return; + + scan_id3_tag(id3_tag, handler, handler_ctx); + + id3_tag_delete(id3_tag); + + return; +} +#endif diff --git a/src/decoder/DsdLib.hxx b/src/decoder/DsdLib.hxx new file mode 100644 index 000000000..2a8e15190 --- /dev/null +++ b/src/decoder/DsdLib.hxx @@ -0,0 +1,51 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_DSDLIB_HXX +#define MPD_DECODER_DSDLIB_HXX + +#include <stdlib.h> + +#include <glib.h> + +struct dsdlib_id { + char value[4]; +}; + +bool +dsdlib_id_equals(const struct dsdlib_id *id, const char *s); + +bool +dsdlib_read(struct decoder *decoder, struct input_stream *is, + void *data, size_t length); + +bool +dsdlib_skip_to(struct decoder *decoder, struct input_stream *is, + goffset offset); + +bool +dsdlib_skip(struct decoder *decoder, struct input_stream *is, + goffset delta); + +void +dsdlib_tag_id3(struct input_stream *is, + const struct tag_handler *handler, + void *handler_ctx, goffset tagoffset); + +#endif diff --git a/src/decoder/DsdiffDecoderPlugin.cxx b/src/decoder/DsdiffDecoderPlugin.cxx new file mode 100644 index 000000000..80b88a2c2 --- /dev/null +++ b/src/decoder/DsdiffDecoderPlugin.cxx @@ -0,0 +1,529 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* \file + * + * This plugin decodes DSDIFF data (SACD) embedded in DFF files. + * The DFF code was modeled after the specification found here: + * http://www.sonicstudio.com/pdf/dsd/DSDIFF_1.5_Spec.pdf + * + * All functions common to both DSD decoders have been moved to dsdlib + */ + +#include "config.h" +#include "DsdiffDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "util/bit_reverse.h" +#include "util/Error.hxx" +#include "tag/TagHandler.hxx" +#include "DsdLib.hxx" + +#include <unistd.h> +#include <stdio.h> /* for SEEK_SET, SEEK_CUR */ + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "dsdiff" + +struct DsdiffHeader { + struct dsdlib_id id; + uint32_t size_high, size_low; + struct dsdlib_id format; +}; + +struct DsdiffChunkHeader { + struct dsdlib_id id; + uint32_t size_high, size_low; + + /** + * Read the "size" attribute from the specified header, converting it + * to the host byte order if needed. + */ + gcc_const + uint64_t GetSize() const { + return (((uint64_t)GUINT32_FROM_BE(size_high)) << 32) | + ((uint64_t)GUINT32_FROM_BE(size_low)); + } +}; + +/** struct for DSDIFF native Artist and Title tags */ +struct dsdiff_native_tag { + uint32_t size; +}; + +struct DsdiffMetaData { + unsigned sample_rate, channels; + bool bitreverse; + uint64_t chunk_size; +#ifdef HAVE_ID3TAG + goffset id3_offset; + uint64_t id3_size; +#endif + /** offset for artist tag */ + goffset diar_offset; + /** offset for title tag */ + goffset diti_offset; +}; + +static bool lsbitfirst; + +static bool +dsdiff_init(const config_param ¶m) +{ + lsbitfirst = param.GetBlockValue("lsbitfirst", false); + return true; +} + +static bool +dsdiff_read_id(struct decoder *decoder, struct input_stream *is, + struct dsdlib_id *id) +{ + return dsdlib_read(decoder, is, id, sizeof(*id)); +} + +static bool +dsdiff_read_chunk_header(struct decoder *decoder, struct input_stream *is, + DsdiffChunkHeader *header) +{ + return dsdlib_read(decoder, is, header, sizeof(*header)); +} + +static bool +dsdiff_read_payload(struct decoder *decoder, struct input_stream *is, + const DsdiffChunkHeader *header, + void *data, size_t length) +{ + uint64_t size = header->GetSize(); + if (size != (uint64_t)length) + return false; + + size_t nbytes = decoder_read(decoder, is, data, length); + return nbytes == length; +} + +/** + * Read and parse a "SND" chunk inside "PROP". + */ +static bool +dsdiff_read_prop_snd(struct decoder *decoder, struct input_stream *is, + DsdiffMetaData *metadata, + goffset end_offset) +{ + DsdiffChunkHeader header; + while ((goffset)(is->GetOffset() + sizeof(header)) <= end_offset) { + if (!dsdiff_read_chunk_header(decoder, is, &header)) + return false; + + goffset chunk_end_offset = is->GetOffset() + + header.GetSize(); + if (chunk_end_offset > end_offset) + return false; + + if (dsdlib_id_equals(&header.id, "FS ")) { + uint32_t sample_rate; + if (!dsdiff_read_payload(decoder, is, &header, + &sample_rate, + sizeof(sample_rate))) + return false; + + metadata->sample_rate = GUINT32_FROM_BE(sample_rate); + } else if (dsdlib_id_equals(&header.id, "CHNL")) { + uint16_t channels; + if (header.GetSize() < sizeof(channels) || + !dsdlib_read(decoder, is, + &channels, sizeof(channels)) || + !dsdlib_skip_to(decoder, is, chunk_end_offset)) + return false; + + metadata->channels = GUINT16_FROM_BE(channels); + } else if (dsdlib_id_equals(&header.id, "CMPR")) { + struct dsdlib_id type; + if (header.GetSize() < sizeof(type) || + !dsdlib_read(decoder, is, + &type, sizeof(type)) || + !dsdlib_skip_to(decoder, is, chunk_end_offset)) + return false; + + if (!dsdlib_id_equals(&type, "DSD ")) + /* only uncompressed DSD audio data + is implemented */ + return false; + } else { + /* ignore unknown chunk */ + + if (!dsdlib_skip_to(decoder, is, chunk_end_offset)) + return false; + } + } + + return is->GetOffset() == end_offset; +} + +/** + * Read and parse a "PROP" chunk. + */ +static bool +dsdiff_read_prop(struct decoder *decoder, struct input_stream *is, + DsdiffMetaData *metadata, + const DsdiffChunkHeader *prop_header) +{ + uint64_t prop_size = prop_header->GetSize(); + goffset end_offset = is->GetOffset() + prop_size; + + struct dsdlib_id prop_id; + if (prop_size < sizeof(prop_id) || + !dsdiff_read_id(decoder, is, &prop_id)) + return false; + + if (dsdlib_id_equals(&prop_id, "SND ")) + return dsdiff_read_prop_snd(decoder, is, metadata, end_offset); + else + /* ignore unknown PROP chunk */ + return dsdlib_skip_to(decoder, is, end_offset); +} + +static void +dsdiff_handle_native_tag(struct input_stream *is, + const struct tag_handler *handler, + void *handler_ctx, goffset tagoffset, + enum tag_type type) +{ + if (!dsdlib_skip_to(nullptr, is, tagoffset)) + return; + + struct dsdiff_native_tag metatag; + + if (!dsdlib_read(nullptr, is, &metatag, sizeof(metatag))) + return; + + uint32_t length = GUINT32_FROM_BE(metatag.size); + + /* Check and limit size of the tag to prevent a stack overflow */ + if (length == 0 || length > 60) + return; + + char string[length]; + char *label; + label = string; + + if (!dsdlib_read(nullptr, is, label, (size_t)length)) + return; + + string[length] = '\0'; + tag_handler_invoke_tag(handler, handler_ctx, type, label); + return; +} + +/** + * Read and parse additional metadata chunks for tagging purposes. By default + * dsdiff files only support equivalents for artist and title but some of the + * extract tools add an id3 tag to provide more tags. If such id3 is found + * this will be used for tagging otherwise the native tags (if any) will be + * used + */ + +static bool +dsdiff_read_metadata_extra(struct decoder *decoder, struct input_stream *is, + DsdiffMetaData *metadata, + DsdiffChunkHeader *chunk_header, + const struct tag_handler *handler, + void *handler_ctx) +{ + + /* skip from DSD data to next chunk header */ + if (!dsdlib_skip(decoder, is, metadata->chunk_size)) + return false; + if (!dsdiff_read_chunk_header(decoder, is, chunk_header)) + return false; + +#ifdef HAVE_ID3TAG + metadata->id3_size = 0; +#endif + + /* Now process all the remaining chunk headers in the stream + and record their position and size */ + + const goffset size = is->GetSize(); + while (is->GetOffset() < size) { + uint64_t chunk_size = chunk_header->GetSize(); + + /* DIIN chunk, is directly followed by other chunks */ + if (dsdlib_id_equals(&chunk_header->id, "DIIN")) + chunk_size = 0; + + /* DIAR chunk - DSDIFF native tag for Artist */ + if (dsdlib_id_equals(&chunk_header->id, "DIAR")) { + chunk_size = chunk_header->GetSize(); + metadata->diar_offset = is->GetOffset(); + } + + /* DITI chunk - DSDIFF native tag for Title */ + if (dsdlib_id_equals(&chunk_header->id, "DITI")) { + chunk_size = chunk_header->GetSize(); + metadata->diti_offset = is->GetOffset(); + } +#ifdef HAVE_ID3TAG + /* 'ID3 ' chunk, offspec. Used by sacdextract */ + if (dsdlib_id_equals(&chunk_header->id, "ID3 ")) { + chunk_size = chunk_header->GetSize(); + metadata->id3_offset = is->GetOffset(); + metadata->id3_size = chunk_size; + } +#endif + if (chunk_size != 0) { + if (!dsdlib_skip(decoder, is, chunk_size)) + break; + } + + if (is->GetOffset() < size) { + if (!dsdiff_read_chunk_header(decoder, is, chunk_header)) + return false; + } + chunk_size = 0; + } + /* done processing chunk headers, process tags if any */ + +#ifdef HAVE_ID3TAG + if (metadata->id3_offset != 0) + { + /* a ID3 tag has preference over the other tags, do not process + other tags if we have one */ + dsdlib_tag_id3(is, handler, handler_ctx, metadata->id3_offset); + return true; + } +#endif + + if (metadata->diar_offset != 0) + dsdiff_handle_native_tag(is, handler, handler_ctx, + metadata->diar_offset, TAG_ARTIST); + + if (metadata->diti_offset != 0) + dsdiff_handle_native_tag(is, handler, handler_ctx, + metadata->diti_offset, TAG_TITLE); + return true; +} + +/** + * Read and parse all metadata chunks at the beginning. Stop when the + * first "DSD" chunk is seen, and return its header in the + * "chunk_header" parameter. + */ +static bool +dsdiff_read_metadata(struct decoder *decoder, struct input_stream *is, + DsdiffMetaData *metadata, + DsdiffChunkHeader *chunk_header) +{ + DsdiffHeader header; + if (!dsdlib_read(decoder, is, &header, sizeof(header)) || + !dsdlib_id_equals(&header.id, "FRM8") || + !dsdlib_id_equals(&header.format, "DSD ")) + return false; + + while (true) { + if (!dsdiff_read_chunk_header(decoder, is, + chunk_header)) + return false; + + if (dsdlib_id_equals(&chunk_header->id, "PROP")) { + if (!dsdiff_read_prop(decoder, is, metadata, + chunk_header)) + return false; + } else if (dsdlib_id_equals(&chunk_header->id, "DSD ")) { + const uint64_t chunk_size = chunk_header->GetSize(); + metadata->chunk_size = chunk_size; + return true; + } else { + /* ignore unknown chunk */ + const uint64_t chunk_size = chunk_header->GetSize(); + goffset chunk_end_offset = is->GetOffset() + + chunk_size; + + if (!dsdlib_skip_to(decoder, is, chunk_end_offset)) + return false; + } + } +} + +static void +bit_reverse_buffer(uint8_t *p, uint8_t *end) +{ + for (; p < end; ++p) + *p = bit_reverse(*p); +} + +/** + * Decode one "DSD" chunk. + */ +static bool +dsdiff_decode_chunk(struct decoder *decoder, struct input_stream *is, + unsigned channels, + uint64_t chunk_size) +{ + uint8_t buffer[8192]; + + const size_t sample_size = sizeof(buffer[0]); + const size_t frame_size = channels * sample_size; + const unsigned buffer_frames = sizeof(buffer) / frame_size; + const unsigned buffer_samples = buffer_frames * frame_size; + const size_t buffer_size = buffer_samples * sample_size; + + while (chunk_size > 0) { + /* see how much aligned data from the remaining chunk + fits into the local buffer */ + unsigned now_frames = buffer_frames; + size_t now_size = buffer_size; + if (chunk_size < (uint64_t)now_size) { + now_frames = (unsigned)chunk_size / frame_size; + now_size = now_frames * frame_size; + } + + size_t nbytes = decoder_read(decoder, is, buffer, now_size); + if (nbytes != now_size) + return false; + + chunk_size -= nbytes; + + if (lsbitfirst) + bit_reverse_buffer(buffer, buffer + nbytes); + + const auto cmd = decoder_data(decoder, is, buffer, nbytes, 0); + switch (cmd) { + case DecoderCommand::NONE: + break; + + case DecoderCommand::START: + case DecoderCommand::STOP: + return false; + + case DecoderCommand::SEEK: + + /* Not implemented yet */ + decoder_seek_error(decoder); + break; + } + } + return dsdlib_skip(decoder, is, chunk_size); +} + +static void +dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is) +{ + DsdiffMetaData metadata; + + DsdiffChunkHeader chunk_header; + /* check if it is is a proper DFF file */ + if (!dsdiff_read_metadata(decoder, is, &metadata, &chunk_header)) + return; + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, + metadata.channels, error)) { + g_warning("%s", error.GetMessage()); + return; + } + + /* calculate song time from DSD chunk size and sample frequency */ + uint64_t chunk_size = metadata.chunk_size; + float songtime = ((chunk_size / metadata.channels) * 8) / + (float) metadata.sample_rate; + + /* success: file was recognized */ + decoder_initialized(decoder, audio_format, false, songtime); + + /* every iteration of the following loop decodes one "DSD" + chunk from a DFF file */ + + while (true) { + chunk_size = chunk_header.GetSize(); + + if (dsdlib_id_equals(&chunk_header.id, "DSD ")) { + if (!dsdiff_decode_chunk(decoder, is, + metadata.channels, + chunk_size)) + break; + } else { + /* ignore other chunks */ + if (!dsdlib_skip(decoder, is, chunk_size)) + break; + } + + /* read next chunk header; the first one was read by + dsdiff_read_metadata() */ + if (!dsdiff_read_chunk_header(decoder, + is, &chunk_header)) + break; + } +} + +static bool +dsdiff_scan_stream(struct input_stream *is, + gcc_unused const struct tag_handler *handler, + gcc_unused void *handler_ctx) +{ + DsdiffMetaData metadata; + DsdiffChunkHeader chunk_header; + + /* First check for DFF metadata */ + if (!dsdiff_read_metadata(nullptr, is, &metadata, &chunk_header)) + return false; + + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, + metadata.channels, IgnoreError())) + /* refuse to parse files which we cannot play anyway */ + return false; + + /* calculate song time and add as tag */ + unsigned songtime = ((metadata.chunk_size / metadata.channels) * 8) / + metadata.sample_rate; + tag_handler_invoke_duration(handler, handler_ctx, songtime); + + /* Read additional metadata and created tags if available */ + dsdiff_read_metadata_extra(nullptr, is, &metadata, &chunk_header, + handler, handler_ctx); + + return true; +} + +static const char *const dsdiff_suffixes[] = { + "dff", + nullptr +}; + +static const char *const dsdiff_mime_types[] = { + "application/x-dff", + nullptr +}; + +const struct decoder_plugin dsdiff_decoder_plugin = { + "dsdiff", + dsdiff_init, + nullptr, + dsdiff_stream_decode, + nullptr, + nullptr, + dsdiff_scan_stream, + nullptr, + dsdiff_suffixes, + dsdiff_mime_types, +}; diff --git a/src/decoder/DsdiffDecoderPlugin.hxx b/src/decoder/DsdiffDecoderPlugin.hxx new file mode 100644 index 000000000..c50605457 --- /dev/null +++ b/src/decoder/DsdiffDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_DSDIFF_H +#define MPD_DECODER_DSDIFF_H + +extern const struct decoder_plugin dsdiff_decoder_plugin; + +#endif diff --git a/src/decoder/DsfDecoderPlugin.cxx b/src/decoder/DsfDecoderPlugin.cxx new file mode 100644 index 000000000..b327fc9dc --- /dev/null +++ b/src/decoder/DsfDecoderPlugin.cxx @@ -0,0 +1,360 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* \file + * + * This plugin decodes DSDIFF data (SACD) embedded in DSF files. + * + * The DSF code was created using the specification found here: + * http://dsd-guide.com/sonys-dsf-file-format-spec + * + * All functions common to both DSD decoders have been moved to dsdlib + */ + +#include "config.h" +#include "DsfDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "util/bit_reverse.h" +#include "util/Error.hxx" +#include "DsdLib.hxx" +#include "tag/TagHandler.hxx" + +#include <unistd.h> +#include <stdio.h> /* for SEEK_SET, SEEK_CUR */ + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "dsf" + +struct DsfMetaData { + unsigned sample_rate, channels; + bool bitreverse; + uint64_t chunk_size; +#ifdef HAVE_ID3TAG + goffset id3_offset; + uint64_t id3_size; +#endif +}; + +struct DsfHeader { + /** DSF header id: "DSD " */ + struct dsdlib_id id; + /** DSD chunk size, including id = 28 */ + uint32_t size_low, size_high; + /** total file size */ + uint32_t fsize_low, fsize_high; + /** pointer to id3v2 metadata, should be at the end of the file */ + uint32_t pmeta_low, pmeta_high; +}; + +/** DSF file fmt chunk */ +struct DsfFmtChunk { + /** id: "fmt " */ + struct dsdlib_id id; + /** fmt chunk size, including id, normally 52 */ + uint32_t size_low, size_high; + /** version of this format = 1 */ + uint32_t version; + /** 0: DSD raw */ + uint32_t formatid; + /** channel type, 1 = mono, 2 = stereo, 3 = 3 channels, etc */ + uint32_t channeltype; + /** Channel number, 1 = mono, 2 = stereo, ... 6 = 6 channels */ + uint32_t channelnum; + /** sample frequency: 2822400, 5644800 */ + uint32_t sample_freq; + /** bits per sample 1 or 8 */ + uint32_t bitssample; + /** Sample count per channel in bytes */ + uint32_t scnt_low, scnt_high; + /** block size per channel = 4096 */ + uint32_t block_size; + /** reserved, should be all zero */ + uint32_t reserved; +}; + +struct DsfDataChunk { + struct dsdlib_id id; + /** "data" chunk size, includes header (id+size) */ + uint32_t size_low, size_high; +}; + +/** + * Read and parse all needed metadata chunks for DSF files. + */ +static bool +dsf_read_metadata(struct decoder *decoder, struct input_stream *is, + DsfMetaData *metadata) +{ + uint64_t chunk_size; + DsfHeader dsf_header; + if (!dsdlib_read(decoder, is, &dsf_header, sizeof(dsf_header)) || + !dsdlib_id_equals(&dsf_header.id, "DSD ")) + return false; + + chunk_size = (((uint64_t)GUINT32_FROM_LE(dsf_header.size_high)) << 32) | + ((uint64_t)GUINT32_FROM_LE(dsf_header.size_low)); + + if (sizeof(dsf_header) != chunk_size) + return false; + +#ifdef HAVE_ID3TAG + uint64_t metadata_offset; + metadata_offset = (((uint64_t)GUINT32_FROM_LE(dsf_header.pmeta_high)) << 32) | + ((uint64_t)GUINT32_FROM_LE(dsf_header.pmeta_low)); +#endif + + /* read the 'fmt ' chunk of the DSF file */ + DsfFmtChunk dsf_fmt_chunk; + if (!dsdlib_read(decoder, is, &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) || + !dsdlib_id_equals(&dsf_fmt_chunk.id, "fmt ")) + return false; + + uint64_t fmt_chunk_size; + fmt_chunk_size = (((uint64_t)GUINT32_FROM_LE(dsf_fmt_chunk.size_high)) << 32) | + ((uint64_t)GUINT32_FROM_LE(dsf_fmt_chunk.size_low)); + + if (fmt_chunk_size != sizeof(dsf_fmt_chunk)) + return false; + + uint32_t samplefreq = (uint32_t)GUINT32_FROM_LE(dsf_fmt_chunk.sample_freq); + + /* for now, only support version 1 of the standard, DSD raw stereo + files with a sample freq of 2822400 Hz */ + + if (dsf_fmt_chunk.version != 1 || dsf_fmt_chunk.formatid != 0 + || dsf_fmt_chunk.channeltype != 2 + || dsf_fmt_chunk.channelnum != 2 + || samplefreq != 2822400) + return false; + + uint32_t chblksize = (uint32_t)GUINT32_FROM_LE(dsf_fmt_chunk.block_size); + /* according to the spec block size should always be 4096 */ + if (chblksize != 4096) + return false; + + /* read the 'data' chunk of the DSF file */ + DsfDataChunk data_chunk; + if (!dsdlib_read(decoder, is, &data_chunk, sizeof(data_chunk)) || + !dsdlib_id_equals(&data_chunk.id, "data")) + return false; + + /* data size of DSF files are padded to multiple of 4096, + we use the actual data size as chunk size */ + + uint64_t data_size; + data_size = (((uint64_t)GUINT32_FROM_LE(data_chunk.size_high)) << 32) | + ((uint64_t)GUINT32_FROM_LE(data_chunk.size_low)); + data_size -= sizeof(data_chunk); + + metadata->chunk_size = data_size; + /* data_size cannot be bigger or equal to total file size */ + const uint64_t size = (uint64_t)is->GetSize(); + if (data_size >= size) + return false; + + metadata->channels = (unsigned) dsf_fmt_chunk.channelnum; + metadata->sample_rate = samplefreq; +#ifdef HAVE_ID3TAG + /* metada_offset cannot be bigger then or equal to total file size */ + if (metadata_offset >= size) + metadata->id3_offset = 0; + else + metadata->id3_offset = (goffset) metadata_offset; +#endif + /* check bits per sample format, determine if bitreverse is needed */ + metadata->bitreverse = dsf_fmt_chunk.bitssample == 1; + return true; +} + +static void +bit_reverse_buffer(uint8_t *p, uint8_t *end) +{ + for (; p < end; ++p) + *p = bit_reverse(*p); +} + +/** + * DSF data is build up of alternating 4096 blocks of DSD samples for left and + * right. Convert the buffer holding 1 block of 4096 DSD left samples and 1 + * block of 4096 DSD right samples to 8k of samples in normal PCM left/right + * order. + */ +static void +dsf_to_pcm_order(uint8_t *dest, uint8_t *scratch, size_t nrbytes) +{ + for (unsigned i = 0, j = 0; i < (unsigned)nrbytes; i += 2) { + scratch[i] = *(dest+j); + j++; + } + + for (unsigned i = 1, j = 0; i < (unsigned) nrbytes; i += 2) { + scratch[i] = *(dest+4096+j); + j++; + } + + for (unsigned i = 0; i < (unsigned)nrbytes; i++) { + *dest = scratch[i]; + dest++; + } +} + +/** + * Decode one complete DSF 'data' chunk i.e. a complete song + */ +static bool +dsf_decode_chunk(struct decoder *decoder, struct input_stream *is, + unsigned channels, + uint64_t chunk_size, + bool bitreverse) +{ + uint8_t buffer[8192]; + + /* scratch buffer for DSF samples to convert to the needed + normal left/right regime of samples */ + uint8_t dsf_scratch_buffer[8192]; + + const size_t sample_size = sizeof(buffer[0]); + const size_t frame_size = channels * sample_size; + const unsigned buffer_frames = sizeof(buffer) / frame_size; + const unsigned buffer_samples = buffer_frames * frame_size; + const size_t buffer_size = buffer_samples * sample_size; + + while (chunk_size > 0) { + /* see how much aligned data from the remaining chunk + fits into the local buffer */ + unsigned now_frames = buffer_frames; + size_t now_size = buffer_size; + if (chunk_size < (uint64_t)now_size) { + now_frames = (unsigned)chunk_size / frame_size; + now_size = now_frames * frame_size; + } + + size_t nbytes = decoder_read(decoder, is, buffer, now_size); + if (nbytes != now_size) + return false; + + chunk_size -= nbytes; + + if (bitreverse) + bit_reverse_buffer(buffer, buffer + nbytes); + + dsf_to_pcm_order(buffer, dsf_scratch_buffer, nbytes); + + const auto cmd = decoder_data(decoder, is, buffer, nbytes, 0); + switch (cmd) { + case DecoderCommand::NONE: + break; + + case DecoderCommand::START: + case DecoderCommand::STOP: + return false; + + case DecoderCommand::SEEK: + + /* not implemented yet */ + decoder_seek_error(decoder); + break; + } + } + return dsdlib_skip(decoder, is, chunk_size); +} + +static void +dsf_stream_decode(struct decoder *decoder, struct input_stream *is) +{ + /* check if it is a proper DSF file */ + DsfMetaData metadata; + if (!dsf_read_metadata(decoder, is, &metadata)) + return; + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, + metadata.channels, error)) { + g_warning("%s", error.GetMessage()); + return; + } + /* Calculate song time from DSD chunk size and sample frequency */ + uint64_t chunk_size = metadata.chunk_size; + float songtime = ((chunk_size / metadata.channels) * 8) / + (float) metadata.sample_rate; + + /* success: file was recognized */ + decoder_initialized(decoder, audio_format, false, songtime); + + if (!dsf_decode_chunk(decoder, is, metadata.channels, + chunk_size, + metadata.bitreverse)) + return; +} + +static bool +dsf_scan_stream(struct input_stream *is, + gcc_unused const struct tag_handler *handler, + gcc_unused void *handler_ctx) +{ + /* check DSF metadata */ + DsfMetaData metadata; + if (!dsf_read_metadata(NULL, is, &metadata)) + return false; + + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8, + SampleFormat::DSD, + metadata.channels, IgnoreError())) + /* refuse to parse files which we cannot play anyway */ + return false; + + /* calculate song time and add as tag */ + unsigned songtime = ((metadata.chunk_size / metadata.channels) * 8) / + metadata.sample_rate; + tag_handler_invoke_duration(handler, handler_ctx, songtime); + +#ifdef HAVE_ID3TAG + /* Add available tags from the ID3 tag */ + dsdlib_tag_id3(is, handler, handler_ctx, metadata.id3_offset); +#endif + return true; +} + +static const char *const dsf_suffixes[] = { + "dsf", + NULL +}; + +static const char *const dsf_mime_types[] = { + "application/x-dsf", + NULL +}; + +const struct decoder_plugin dsf_decoder_plugin = { + "dsf", + nullptr, + nullptr, + dsf_stream_decode, + nullptr, + nullptr, + dsf_scan_stream, + nullptr, + dsf_suffixes, + dsf_mime_types, +}; diff --git a/src/decoder/DsfDecoderPlugin.hxx b/src/decoder/DsfDecoderPlugin.hxx new file mode 100644 index 000000000..749032d1f --- /dev/null +++ b/src/decoder/DsfDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_DSF_H +#define MPD_DECODER_DSF_H + +extern const struct decoder_plugin dsf_decoder_plugin; + +#endif diff --git a/src/decoder/FaadDecoderPlugin.cxx b/src/decoder/FaadDecoderPlugin.cxx new file mode 100644 index 000000000..f026a6216 --- /dev/null +++ b/src/decoder/FaadDecoderPlugin.cxx @@ -0,0 +1,496 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "FaadDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "DecoderBuffer.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "util/Error.hxx" +#include "util/Domain.hxx" + +#include <neaacdec.h> + +#include <glib.h> + +#include <assert.h> +#include <string.h> +#include <unistd.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "faad" + +#define AAC_MAX_CHANNELS 6 + +static const unsigned adts_sample_rates[] = + { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, + 16000, 12000, 11025, 8000, 7350, 0, 0, 0 +}; + +static constexpr Domain faad_decoder_domain("faad_decoder"); + +/** + * Check whether the buffer head is an AAC frame, and return the frame + * length. Returns 0 if it is not a frame. + */ +static size_t +adts_check_frame(const unsigned char *data) +{ + /* check syncword */ + if (!((data[0] == 0xFF) && ((data[1] & 0xF6) == 0xF0))) + return 0; + + return (((unsigned int)data[3] & 0x3) << 11) | + (((unsigned int)data[4]) << 3) | + (data[5] >> 5); +} + +/** + * Find the next AAC frame in the buffer. Returns 0 if no frame is + * found or if not enough data is available. + */ +static size_t +adts_find_frame(DecoderBuffer *buffer) +{ + size_t length, frame_length; + bool ret; + + while (true) { + const uint8_t *data = (const uint8_t *) + decoder_buffer_read(buffer, &length); + if (data == nullptr || length < 8) { + /* not enough data yet */ + ret = decoder_buffer_fill(buffer); + if (!ret) + /* failed */ + return 0; + + continue; + } + + /* find the 0xff marker */ + const uint8_t *p = (const uint8_t *)memchr(data, 0xff, length); + if (p == nullptr) { + /* no marker - discard the buffer */ + decoder_buffer_consume(buffer, length); + continue; + } + + if (p > data) { + /* discard data before 0xff */ + decoder_buffer_consume(buffer, p - data); + continue; + } + + /* is it a frame? */ + frame_length = adts_check_frame(data); + if (frame_length == 0) { + /* it's just some random 0xff byte; discard it + and continue searching */ + decoder_buffer_consume(buffer, 1); + continue; + } + + if (length < frame_length) { + /* available buffer size is smaller than the + frame will be - attempt to read more + data */ + ret = decoder_buffer_fill(buffer); + if (!ret) { + /* not enough data; discard this frame + to prevent a possible buffer + overflow */ + data = (const uint8_t *) + decoder_buffer_read(buffer, &length); + if (data != nullptr) + decoder_buffer_consume(buffer, length); + } + + continue; + } + + /* found a full frame! */ + return frame_length; + } +} + +static float +adts_song_duration(DecoderBuffer *buffer) +{ + unsigned int frames, frame_length; + unsigned sample_rate = 0; + float frames_per_second; + + /* Read all frames to ensure correct time and bitrate */ + for (frames = 0;; frames++) { + frame_length = adts_find_frame(buffer); + if (frame_length == 0) + break; + + + if (frames == 0) { + size_t buffer_length; + const uint8_t *data = (const uint8_t *) + decoder_buffer_read(buffer, &buffer_length); + assert(data != nullptr); + assert(frame_length <= buffer_length); + + sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2]; + } + + decoder_buffer_consume(buffer, frame_length); + } + + frames_per_second = (float)sample_rate / 1024.0; + if (frames_per_second <= 0) + return -1; + + return (float)frames / frames_per_second; +} + +static float +faad_song_duration(DecoderBuffer *buffer, struct input_stream *is) +{ + size_t fileread; + size_t tagsize; + size_t length; + bool success; + + const goffset size = is->GetSize(); + fileread = size >= 0 ? size : 0; + + decoder_buffer_fill(buffer); + const uint8_t *data = (const uint8_t *) + decoder_buffer_read(buffer, &length); + if (data == nullptr) + return -1; + + tagsize = 0; + if (length >= 10 && !memcmp(data, "ID3", 3)) { + /* skip the ID3 tag */ + + tagsize = (data[6] << 21) | (data[7] << 14) | + (data[8] << 7) | (data[9] << 0); + + tagsize += 10; + + success = decoder_buffer_skip(buffer, tagsize) && + decoder_buffer_fill(buffer); + if (!success) + return -1; + + data = (const uint8_t *)decoder_buffer_read(buffer, &length); + if (data == nullptr) + return -1; + } + + if (is->IsSeekable() && length >= 2 && + data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) { + /* obtain the duration from the ADTS header */ + float song_length = adts_song_duration(buffer); + + is->LockSeek(tagsize, SEEK_SET, IgnoreError()); + + data = (const uint8_t *)decoder_buffer_read(buffer, &length); + if (data != nullptr) + decoder_buffer_consume(buffer, length); + decoder_buffer_fill(buffer); + + return song_length; + } else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) { + /* obtain the duration from the ADIF header */ + unsigned bit_rate; + size_t skip_size = (data[4] & 0x80) ? 9 : 0; + + if (8 + skip_size > length) + /* not enough data yet; skip parsing this + header */ + return -1; + + bit_rate = ((data[4 + skip_size] & 0x0F) << 19) | + (data[5 + skip_size] << 11) | + (data[6 + skip_size] << 3) | + (data[7 + skip_size] & 0xE0); + + if (fileread != 0 && bit_rate != 0) + return fileread * 8.0 / bit_rate; + else + return fileread; + } else + return -1; +} + +/** + * Wrapper for NeAACDecInit() which works around some API + * inconsistencies in libfaad. + */ +static bool +faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer, + AudioFormat &audio_format, Error &error) +{ + int32_t nbytes; + uint32_t sample_rate; + uint8_t channels; +#ifdef HAVE_FAAD_LONG + /* neaacdec.h declares all arguments as "unsigned long", but + internally expects uint32_t pointers. To avoid gcc + warnings, use this workaround. */ + unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate; +#else + uint32_t *sample_rate_p = &sample_rate; +#endif + + size_t length; + const unsigned char *data = (const unsigned char *) + decoder_buffer_read(buffer, &length); + if (data == nullptr) { + error.Set(faad_decoder_domain, "Empty file"); + return false; + } + + nbytes = NeAACDecInit(decoder, + /* deconst hack, libfaad requires this */ + const_cast<unsigned char *>(data), + length, + sample_rate_p, &channels); + if (nbytes < 0) { + error.Set(faad_decoder_domain, "Not an AAC stream"); + return false; + } + + decoder_buffer_consume(buffer, nbytes); + + return audio_format_init_checked(audio_format, sample_rate, + SampleFormat::S16, channels, error); +} + +/** + * Wrapper for NeAACDecDecode() which works around some API + * inconsistencies in libfaad. + */ +static const void * +faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer *buffer, + NeAACDecFrameInfo *frame_info) +{ + size_t length; + const unsigned char *data = (const unsigned char *) + decoder_buffer_read(buffer, &length); + if (data == nullptr) + return nullptr; + + return NeAACDecDecode(decoder, frame_info, + /* deconst hack, libfaad requires this */ + const_cast<unsigned char *>(data), + length); +} + +/** + * Get a song file's total playing time in seconds, as a float. + * Returns 0 if the duration is unknown, and a negative value if the + * file is invalid. + */ +static float +faad_get_file_time_float(struct input_stream *is) +{ + DecoderBuffer *buffer; + float length; + + buffer = decoder_buffer_new(nullptr, is, + FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); + length = faad_song_duration(buffer, is); + + if (length < 0) { + bool ret; + AudioFormat audio_format; + + NeAACDecHandle decoder = NeAACDecOpen(); + + NeAACDecConfigurationPtr config = + NeAACDecGetCurrentConfiguration(decoder); + config->outputFormat = FAAD_FMT_16BIT; + NeAACDecSetConfiguration(decoder, config); + + decoder_buffer_fill(buffer); + + ret = faad_decoder_init(decoder, buffer, audio_format, + IgnoreError()); + if (ret) + length = 0; + + NeAACDecClose(decoder); + } + + decoder_buffer_free(buffer); + + return length; +} + +/** + * Get a song file's total playing time in seconds, as an int. + * Returns 0 if the duration is unknown, and a negative value if the + * file is invalid. + */ +static int +faad_get_file_time(struct input_stream *is) +{ + int file_time = -1; + float length; + + if ((length = faad_get_file_time_float(is)) >= 0) + file_time = length + 0.5; + + return file_time; +} + +static void +faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) +{ + float total_time = 0; + AudioFormat audio_format; + bool ret; + uint16_t bit_rate = 0; + DecoderBuffer *buffer; + + buffer = decoder_buffer_new(mpd_decoder, is, + FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); + total_time = faad_song_duration(buffer, is); + + /* create the libfaad decoder */ + + NeAACDecHandle decoder = NeAACDecOpen(); + + NeAACDecConfigurationPtr config = + NeAACDecGetCurrentConfiguration(decoder); + config->outputFormat = FAAD_FMT_16BIT; + config->downMatrix = 1; + config->dontUpSampleImplicitSBR = 0; + NeAACDecSetConfiguration(decoder, config); + + while (!decoder_buffer_is_full(buffer) && !is->LockIsEOF() && + decoder_get_command(mpd_decoder) == DecoderCommand::NONE) { + adts_find_frame(buffer); + decoder_buffer_fill(buffer); + } + + /* initialize it */ + + Error error; + ret = faad_decoder_init(decoder, buffer, audio_format, error); + if (!ret) { + g_warning("%s", error.GetMessage()); + NeAACDecClose(decoder); + return; + } + + /* initialize the MPD core */ + + decoder_initialized(mpd_decoder, audio_format, false, total_time); + + /* the decoder loop */ + + DecoderCommand cmd; + do { + size_t frame_size; + const void *decoded; + NeAACDecFrameInfo frame_info; + + /* find the next frame */ + + frame_size = adts_find_frame(buffer); + if (frame_size == 0) + /* end of file */ + break; + + /* decode it */ + + decoded = faad_decoder_decode(decoder, buffer, &frame_info); + + if (frame_info.error > 0) { + g_warning("error decoding AAC stream: %s\n", + NeAACDecGetErrorMessage(frame_info.error)); + break; + } + + if (frame_info.channels != audio_format.channels) { + g_warning("channel count changed from %u to %u", + audio_format.channels, frame_info.channels); + break; + } + + if (frame_info.samplerate != audio_format.sample_rate) { + g_warning("sample rate changed from %u to %lu", + audio_format.sample_rate, + (unsigned long)frame_info.samplerate); + break; + } + + decoder_buffer_consume(buffer, frame_info.bytesconsumed); + + /* update bit rate and position */ + + if (frame_info.samples > 0) { + bit_rate = frame_info.bytesconsumed * 8.0 * + frame_info.channels * audio_format.sample_rate / + frame_info.samples / 1000 + 0.5; + } + + /* send PCM samples to MPD */ + + cmd = decoder_data(mpd_decoder, is, decoded, + (size_t)frame_info.samples * 2, + bit_rate); + } while (cmd != DecoderCommand::STOP); + + /* cleanup */ + + NeAACDecClose(decoder); +} + +static bool +faad_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + int file_time = faad_get_file_time(is); + + if (file_time < 0) + return false; + + tag_handler_invoke_duration(handler, handler_ctx, file_time); + return true; +} + +static const char *const faad_suffixes[] = { "aac", nullptr }; +static const char *const faad_mime_types[] = { + "audio/aac", "audio/aacp", nullptr +}; + +const struct decoder_plugin faad_decoder_plugin = { + "faad", + nullptr, + nullptr, + faad_stream_decode, + nullptr, + nullptr, + faad_scan_stream, + nullptr, + faad_suffixes, + faad_mime_types, +}; diff --git a/src/decoder/FaadDecoderPlugin.hxx b/src/decoder/FaadDecoderPlugin.hxx new file mode 100644 index 000000000..162c155ad --- /dev/null +++ b/src/decoder/FaadDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_FAAD_DECODER_PLUGIN_HXX +#define MPD_FAAD_DECODER_PLUGIN_HXX + +extern const struct decoder_plugin faad_decoder_plugin; + +#endif diff --git a/src/decoder/FfmpegDecoderPlugin.cxx b/src/decoder/FfmpegDecoderPlugin.cxx new file mode 100644 index 000000000..a725e1f7d --- /dev/null +++ b/src/decoder/FfmpegDecoderPlugin.cxx @@ -0,0 +1,681 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* necessary because libavutil/common.h uses UINT64_C */ +#define __STDC_CONSTANT_MACROS + +#include "config.h" +#include "FfmpegDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "FfmpegMetaData.hxx" +#include "tag/TagHandler.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "util/Error.hxx" + +#include <glib.h> + +#include <assert.h> +#include <stdio.h> +#include <unistd.h> +#include <stdlib.h> +#include <string.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <unistd.h> + +extern "C" { +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavformat/avio.h> +#include <libavutil/avutil.h> +#include <libavutil/log.h> +#include <libavutil/mathematics.h> +#include <libavutil/dict.h> +} + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "ffmpeg" + +/* suppress the ffmpeg compatibility macro */ +#ifdef SampleFormat +#undef SampleFormat +#endif + +static GLogLevelFlags +level_ffmpeg_to_glib(int level) +{ + if (level <= AV_LOG_FATAL) + return G_LOG_LEVEL_CRITICAL; + + if (level <= AV_LOG_ERROR) + return G_LOG_LEVEL_WARNING; + + if (level <= AV_LOG_INFO) + return G_LOG_LEVEL_MESSAGE; + + return G_LOG_LEVEL_DEBUG; +} + +static void +mpd_ffmpeg_log_callback(gcc_unused void *ptr, int level, + const char *fmt, va_list vl) +{ + const AVClass * cls = NULL; + + if (ptr != NULL) + cls = *(const AVClass *const*)ptr; + + if (cls != NULL) { + char *domain = g_strconcat(G_LOG_DOMAIN, "/", cls->item_name(ptr), NULL); + g_logv(domain, level_ffmpeg_to_glib(level), fmt, vl); + g_free(domain); + } +} + +struct AvioStream { + struct decoder *decoder; + struct input_stream *input; + + AVIOContext *io; + + unsigned char buffer[8192]; + + AvioStream(struct decoder *_decoder, input_stream *_input) + :decoder(_decoder), input(_input), io(nullptr) {} + + ~AvioStream() { + if (io != nullptr) + av_free(io); + } + + bool Open(); +}; + +static int +mpd_ffmpeg_stream_read(void *opaque, uint8_t *buf, int size) +{ + AvioStream *stream = (AvioStream *)opaque; + + return decoder_read(stream->decoder, stream->input, + (void *)buf, size); +} + +static int64_t +mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence) +{ + AvioStream *stream = (AvioStream *)opaque; + + if (whence == AVSEEK_SIZE) + return stream->input->size; + + Error error; + if (!stream->input->LockSeek(pos, whence, error)) + return -1; + + return stream->input->offset; +} + +bool +AvioStream::Open() +{ + io = avio_alloc_context(buffer, sizeof(buffer), + false, this, + mpd_ffmpeg_stream_read, nullptr, + input->seekable + ? mpd_ffmpeg_stream_seek : nullptr); + return io != nullptr; +} + +/** + * API compatibility wrapper for av_open_input_stream() and + * avformat_open_input(). + */ +static int +mpd_ffmpeg_open_input(AVFormatContext **ic_ptr, + AVIOContext *pb, + const char *filename, + AVInputFormat *fmt) +{ + AVFormatContext *context = avformat_alloc_context(); + if (context == NULL) + return AVERROR(ENOMEM); + + context->pb = pb; + *ic_ptr = context; + return avformat_open_input(ic_ptr, filename, fmt, NULL); +} + +static bool +ffmpeg_init(gcc_unused const config_param ¶m) +{ + av_log_set_callback(mpd_ffmpeg_log_callback); + + av_register_all(); + return true; +} + +static int +ffmpeg_find_audio_stream(const AVFormatContext *format_context) +{ + for (unsigned i = 0; i < format_context->nb_streams; ++i) + if (format_context->streams[i]->codec->codec_type == + AVMEDIA_TYPE_AUDIO) + return i; + + return -1; +} + +gcc_const +static double +time_from_ffmpeg(int64_t t, const AVRational time_base) +{ + assert(t != (int64_t)AV_NOPTS_VALUE); + + return (double)av_rescale_q(t, time_base, (AVRational){1, 1024}) + / (double)1024; +} + +gcc_const +static int64_t +time_to_ffmpeg(double t, const AVRational time_base) +{ + return av_rescale_q((int64_t)(t * 1024), (AVRational){1, 1024}, + time_base); +} + +static void +copy_interleave_frame2(uint8_t *dest, uint8_t **src, + unsigned nframes, unsigned nchannels, + unsigned sample_size) +{ + for (unsigned frame = 0; frame < nframes; ++frame) { + for (unsigned channel = 0; channel < nchannels; ++channel) { + memcpy(dest, src[channel] + frame * sample_size, + sample_size); + dest += sample_size; + } + } +} + +/** + * Copy PCM data from a AVFrame to an interleaved buffer. + */ +static int +copy_interleave_frame(const AVCodecContext *codec_context, + const AVFrame *frame, + uint8_t **output_buffer, + uint8_t **global_buffer, int *global_buffer_size) +{ + int plane_size; + const int data_size = + av_samples_get_buffer_size(&plane_size, + codec_context->channels, + frame->nb_samples, + codec_context->sample_fmt, 1); + if (av_sample_fmt_is_planar(codec_context->sample_fmt) && + codec_context->channels > 1) { + if(*global_buffer_size < data_size) { + av_freep(global_buffer); + + *global_buffer = (uint8_t*)av_malloc(data_size); + + if (!*global_buffer) + /* Not enough memory - shouldn't happen */ + return AVERROR(ENOMEM); + *global_buffer_size = data_size; + } + *output_buffer = *global_buffer; + copy_interleave_frame2(*output_buffer, frame->extended_data, + frame->nb_samples, + codec_context->channels, + av_get_bytes_per_sample(codec_context->sample_fmt)); + } else { + *output_buffer = frame->extended_data[0]; + } + + return data_size; +} + +static DecoderCommand +ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, + const AVPacket *packet, + AVCodecContext *codec_context, + const AVRational *time_base, + AVFrame *frame, + uint8_t **buffer, int *buffer_size) +{ + if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE) + decoder_timestamp(decoder, + time_from_ffmpeg(packet->pts, *time_base)); + + AVPacket packet2 = *packet; + + uint8_t *output_buffer; + + DecoderCommand cmd = DecoderCommand::NONE; + while (packet2.size > 0 && cmd == DecoderCommand::NONE) { + int audio_size = 0; + int got_frame = 0; + int len = avcodec_decode_audio4(codec_context, + frame, &got_frame, + &packet2); + if (len >= 0 && got_frame) { + audio_size = copy_interleave_frame(codec_context, + frame, + &output_buffer, + buffer, buffer_size); + if (audio_size < 0) + len = audio_size; + } + + if (len < 0) { + /* if error, we skip the frame */ + g_message("decoding failed, frame skipped\n"); + break; + } + + packet2.data += len; + packet2.size -= len; + + if (audio_size <= 0) + continue; + + cmd = decoder_data(decoder, is, + output_buffer, audio_size, + codec_context->bit_rate / 1000); + } + return cmd; +} + +gcc_const +static SampleFormat +ffmpeg_sample_format(enum AVSampleFormat sample_fmt) +{ + switch (sample_fmt) { + case AV_SAMPLE_FMT_S16: + case AV_SAMPLE_FMT_S16P: + return SampleFormat::S16; + + case AV_SAMPLE_FMT_S32: + case AV_SAMPLE_FMT_S32P: + return SampleFormat::S32; + + case AV_SAMPLE_FMT_FLTP: + return SampleFormat::FLOAT; + + default: + break; + } + + char buffer[64]; + const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer), + sample_fmt); + if (name != NULL) + g_warning("Unsupported libavcodec SampleFormat value: %s (%d)", + name, sample_fmt); + else + g_warning("Unsupported libavcodec SampleFormat value: %d", + sample_fmt); + return SampleFormat::UNDEFINED; +} + +static AVInputFormat * +ffmpeg_probe(struct decoder *decoder, struct input_stream *is) +{ + enum { + BUFFER_SIZE = 16384, + PADDING = 16, + }; + + Error error; + + unsigned char *buffer = (unsigned char *)g_malloc(BUFFER_SIZE); + size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE); + if (nbytes <= PADDING || !is->LockSeek(0, SEEK_SET, error)) { + g_free(buffer); + return NULL; + } + + /* some ffmpeg parsers (e.g. ac3_parser.c) read a few bytes + beyond the declared buffer limit, which makes valgrind + angry; this workaround removes some padding from the buffer + size */ + nbytes -= PADDING; + + AVProbeData avpd; + avpd.buf = buffer; + avpd.buf_size = nbytes; + avpd.filename = is->uri.c_str(); + + AVInputFormat *format = av_probe_input_format(&avpd, true); + g_free(buffer); + + return format; +} + +static void +ffmpeg_decode(struct decoder *decoder, struct input_stream *input) +{ + AVInputFormat *input_format = ffmpeg_probe(decoder, input); + if (input_format == NULL) + return; + + g_debug("detected input format '%s' (%s)", + input_format->name, input_format->long_name); + + AvioStream stream(decoder, input); + if (!stream.Open()) { + g_warning("Failed to open stream"); + return; + } + + //ffmpeg works with ours "fileops" helper + AVFormatContext *format_context = NULL; + if (mpd_ffmpeg_open_input(&format_context, stream.io, + input->uri.c_str(), + input_format) != 0) { + g_warning("Open failed\n"); + return; + } + + const int find_result = + avformat_find_stream_info(format_context, NULL); + if (find_result < 0) { + g_warning("Couldn't find stream info\n"); + avformat_close_input(&format_context); + return; + } + + int audio_stream = ffmpeg_find_audio_stream(format_context); + if (audio_stream == -1) { + g_warning("No audio stream inside\n"); + avformat_close_input(&format_context); + return; + } + + AVStream *av_stream = format_context->streams[audio_stream]; + + AVCodecContext *codec_context = av_stream->codec; + if (codec_context->codec_name[0] != 0) + g_debug("codec '%s'", codec_context->codec_name); + + AVCodec *codec = avcodec_find_decoder(codec_context->codec_id); + + if (!codec) { + g_warning("Unsupported audio codec\n"); + avformat_close_input(&format_context); + return; + } + + const SampleFormat sample_format = + ffmpeg_sample_format(codec_context->sample_fmt); + if (sample_format == SampleFormat::UNDEFINED) + return; + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, + codec_context->sample_rate, + sample_format, + codec_context->channels, error)) { + g_warning("%s", error.GetMessage()); + avformat_close_input(&format_context); + return; + } + + /* the audio format must be read from AVCodecContext by now, + because avcodec_open() has been demonstrated to fill bogus + values into AVCodecContext.channels - a change that will be + reverted later by avcodec_decode_audio3() */ + + const int open_result = avcodec_open2(codec_context, codec, NULL); + if (open_result < 0) { + g_warning("Could not open codec\n"); + avformat_close_input(&format_context); + return; + } + + int total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE + ? format_context->duration / AV_TIME_BASE + : 0; + + decoder_initialized(decoder, audio_format, + input->seekable, total_time); + + AVFrame *frame = avcodec_alloc_frame(); + if (!frame) { + g_warning("Could not allocate frame\n"); + avformat_close_input(&format_context); + return; + } + + uint8_t *interleaved_buffer = NULL; + int interleaved_buffer_size = 0; + + DecoderCommand cmd; + do { + AVPacket packet; + if (av_read_frame(format_context, &packet) < 0) + /* end of file */ + break; + + if (packet.stream_index == audio_stream) + cmd = ffmpeg_send_packet(decoder, input, + &packet, codec_context, + &av_stream->time_base, + frame, + &interleaved_buffer, &interleaved_buffer_size); + else + cmd = decoder_get_command(decoder); + + av_free_packet(&packet); + + if (cmd == DecoderCommand::SEEK) { + int64_t where = + time_to_ffmpeg(decoder_seek_where(decoder), + av_stream->time_base); + + if (av_seek_frame(format_context, audio_stream, where, + AV_TIME_BASE) < 0) + decoder_seek_error(decoder); + else { + avcodec_flush_buffers(codec_context); + decoder_command_finished(decoder); + } + } + } while (cmd != DecoderCommand::STOP); + +#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) + avcodec_free_frame(&frame); +#else + av_freep(&frame); +#endif + av_freep(&interleaved_buffer); + + avcodec_close(codec_context); + avformat_close_input(&format_context); +} + +//no tag reading in ffmpeg, check if playable +static bool +ffmpeg_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + AVInputFormat *input_format = ffmpeg_probe(NULL, is); + if (input_format == NULL) + return false; + + AvioStream stream(nullptr, is); + if (!stream.Open()) + return false; + + AVFormatContext *f = NULL; + if (mpd_ffmpeg_open_input(&f, stream.io, is->uri.c_str(), + input_format) != 0) + return false; + + const int find_result = + avformat_find_stream_info(f, NULL); + if (find_result < 0) { + avformat_close_input(&f); + return false; + } + + if (f->duration != (int64_t)AV_NOPTS_VALUE) + tag_handler_invoke_duration(handler, handler_ctx, + f->duration / AV_TIME_BASE); + + ffmpeg_scan_dictionary(f->metadata, handler, handler_ctx); + int idx = ffmpeg_find_audio_stream(f); + if (idx >= 0) + ffmpeg_scan_dictionary(f->streams[idx]->metadata, + handler, handler_ctx); + + avformat_close_input(&f); + return true; +} + +/** + * A list of extensions found for the formats supported by ffmpeg. + * This list is current as of 02-23-09; To find out if there are more + * supported formats, check the ffmpeg changelog since this date for + * more formats. + */ +static const char *const ffmpeg_suffixes[] = { + "16sv", "3g2", "3gp", "4xm", "8svx", "aa3", "aac", "ac3", "afc", "aif", + "aifc", "aiff", "al", "alaw", "amr", "anim", "apc", "ape", "asf", + "atrac", "au", "aud", "avi", "avm2", "avs", "bap", "bfi", "c93", "cak", + "cin", "cmv", "cpk", "daud", "dct", "divx", "dts", "dv", "dvd", "dxa", + "eac3", "film", "flac", "flc", "fli", "fll", "flx", "flv", "g726", + "gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts", + "m4a", "m4b", "m4v", + "mad", + "mj2", "mjpeg", "mjpg", "mka", "mkv", "mlp", "mm", "mmf", "mov", "mp+", + "mp1", "mp2", "mp3", "mp4", "mpc", "mpeg", "mpg", "mpga", "mpp", "mpu", + "mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv", + "ogx", "oma", "ogg", "omg", "psp", "pva", "qcp", "qt", "r3d", "ra", + "ram", "rl2", "rm", "rmvb", "roq", "rpl", "rvc", "shn", "smk", "snd", + "sol", "son", "spx", "str", "swf", "tgi", "tgq", "tgv", "thp", "ts", + "tsp", "tta", "xa", "xvid", "uv", "uv2", "vb", "vid", "vob", "voc", + "vp6", "vmd", "wav", "webm", "wma", "wmv", "wsaud", "wsvga", "wv", + "wve", + NULL +}; + +static const char *const ffmpeg_mime_types[] = { + "application/flv", + "application/m4a", + "application/mp4", + "application/octet-stream", + "application/ogg", + "application/x-ms-wmz", + "application/x-ms-wmd", + "application/x-ogg", + "application/x-shockwave-flash", + "application/x-shorten", + "audio/8svx", + "audio/16sv", + "audio/aac", + "audio/ac3", + "audio/aiff" + "audio/amr", + "audio/basic", + "audio/flac", + "audio/m4a", + "audio/mp4", + "audio/mpeg", + "audio/musepack", + "audio/ogg", + "audio/qcelp", + "audio/vorbis", + "audio/vorbis+ogg", + "audio/x-8svx", + "audio/x-16sv", + "audio/x-aac", + "audio/x-ac3", + "audio/x-aiff" + "audio/x-alaw", + "audio/x-au", + "audio/x-dca", + "audio/x-eac3", + "audio/x-flac", + "audio/x-gsm", + "audio/x-mace", + "audio/x-matroska", + "audio/x-monkeys-audio", + "audio/x-mpeg", + "audio/x-ms-wma", + "audio/x-ms-wax", + "audio/x-musepack", + "audio/x-ogg", + "audio/x-vorbis", + "audio/x-vorbis+ogg", + "audio/x-pn-realaudio", + "audio/x-pn-multirate-realaudio", + "audio/x-speex", + "audio/x-tta" + "audio/x-voc", + "audio/x-wav", + "audio/x-wma", + "audio/x-wv", + "video/anim", + "video/quicktime", + "video/msvideo", + "video/ogg", + "video/theora", + "video/webm", + "video/x-dv", + "video/x-flv", + "video/x-matroska", + "video/x-mjpeg", + "video/x-mpeg", + "video/x-ms-asf", + "video/x-msvideo", + "video/x-ms-wmv", + "video/x-ms-wvx", + "video/x-ms-wm", + "video/x-ms-wmx", + "video/x-nut", + "video/x-pva", + "video/x-theora", + "video/x-vid", + "video/x-wmv", + "video/x-xvid", + + /* special value for the "ffmpeg" input plugin: all streams by + the "ffmpeg" input plugin shall be decoded by this + plugin */ + "audio/x-mpd-ffmpeg", + + NULL +}; + +const struct decoder_plugin ffmpeg_decoder_plugin = { + "ffmpeg", + ffmpeg_init, + nullptr, + ffmpeg_decode, + nullptr, + nullptr, + ffmpeg_scan_stream, + nullptr, + ffmpeg_suffixes, + ffmpeg_mime_types +}; diff --git a/src/decoder/FfmpegDecoderPlugin.hxx b/src/decoder/FfmpegDecoderPlugin.hxx new file mode 100644 index 000000000..9a637fff0 --- /dev/null +++ b/src/decoder/FfmpegDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_FFMPEG_HXX +#define MPD_DECODER_FFMPEG_HXX + +extern const struct decoder_plugin ffmpeg_decoder_plugin; + +#endif diff --git a/src/decoder/FfmpegMetaData.cxx b/src/decoder/FfmpegMetaData.cxx new file mode 100644 index 000000000..f4b7386ef --- /dev/null +++ b/src/decoder/FfmpegMetaData.cxx @@ -0,0 +1,79 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* necessary because libavutil/common.h uses UINT64_C */ +#define __STDC_CONSTANT_MACROS + +#include "config.h" +#include "FfmpegMetaData.hxx" +#include "tag/TagTable.hxx" +#include "tag/TagHandler.hxx" + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "ffmpeg" + +static const struct tag_table ffmpeg_tags[] = { + { "year", TAG_DATE }, + { "author-sort", TAG_ARTIST_SORT }, + { "album_artist", TAG_ALBUM_ARTIST }, + { "album_artist-sort", TAG_ALBUM_ARTIST_SORT }, + + /* sentinel */ + { NULL, TAG_NUM_OF_ITEM_TYPES } +}; + +static void +ffmpeg_copy_metadata(enum tag_type type, + AVDictionary *m, const char *name, + const struct tag_handler *handler, void *handler_ctx) +{ + AVDictionaryEntry *mt = NULL; + + while ((mt = av_dict_get(m, name, mt, 0)) != NULL) + tag_handler_invoke_tag(handler, handler_ctx, + type, mt->value); +} + +static void +ffmpeg_scan_pairs(AVDictionary *dict, + const struct tag_handler *handler, void *handler_ctx) +{ + AVDictionaryEntry *i = NULL; + + while ((i = av_dict_get(dict, "", i, AV_DICT_IGNORE_SUFFIX)) != NULL) + tag_handler_invoke_pair(handler, handler_ctx, + i->key, i->value); +} + +void +ffmpeg_scan_dictionary(AVDictionary *dict, + const struct tag_handler *handler, void *handler_ctx) +{ + for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) + ffmpeg_copy_metadata(tag_type(i), dict, tag_item_names[i], + handler, handler_ctx); + + for (const struct tag_table *i = ffmpeg_tags; + i->name != NULL; ++i) + ffmpeg_copy_metadata(i->type, dict, i->name, + handler, handler_ctx); + + if (handler->pair != NULL) + ffmpeg_scan_pairs(dict, handler, handler_ctx); +} diff --git a/src/decoder/FfmpegMetaData.hxx b/src/decoder/FfmpegMetaData.hxx new file mode 100644 index 000000000..0fd73df04 --- /dev/null +++ b/src/decoder/FfmpegMetaData.hxx @@ -0,0 +1,40 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_FFMPEG_METADATA_HXX +#define MPD_FFMPEG_METADATA_HXX + +extern "C" { +#include <libavformat/avformat.h> +#include <libavutil/avutil.h> +#include <libavutil/dict.h> +} + +/* suppress the ffmpeg compatibility macro */ +#ifdef SampleFormat +#undef SampleFormat +#endif + +struct tag_handler; + +void +ffmpeg_scan_dictionary(AVDictionary *dict, + const struct tag_handler *handler, void *handler_ctx); + +#endif diff --git a/src/decoder/FlacCommon.cxx b/src/decoder/FlacCommon.cxx new file mode 100644 index 000000000..9f5d81f85 --- /dev/null +++ b/src/decoder/FlacCommon.cxx @@ -0,0 +1,197 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* + * Common data structures and functions used by FLAC and OggFLAC + */ + +#include "config.h" +#include "FlacCommon.hxx" +#include "FlacMetadata.hxx" +#include "FlacPcm.hxx" +#include "CheckAudioFormat.hxx" +#include "util/Error.hxx" + +#include <glib.h> + +#include <assert.h> + +flac_data::flac_data(struct decoder *_decoder, + struct input_stream *_input_stream) + :FlacInput(_input_stream, _decoder), + initialized(false), unsupported(false), + total_frames(0), first_frame(0), next_frame(0), position(0), + decoder(_decoder), input_stream(_input_stream) +{ +} + +static SampleFormat +flac_sample_format(unsigned bits_per_sample) +{ + switch (bits_per_sample) { + case 8: + return SampleFormat::S8; + + case 16: + return SampleFormat::S16; + + case 24: + return SampleFormat::S24_P32; + + case 32: + return SampleFormat::S32; + + default: + return SampleFormat::UNDEFINED; + } +} + +static void +flac_got_stream_info(struct flac_data *data, + const FLAC__StreamMetadata_StreamInfo *stream_info) +{ + if (data->initialized || data->unsupported) + return; + + Error error; + if (!audio_format_init_checked(data->audio_format, + stream_info->sample_rate, + flac_sample_format(stream_info->bits_per_sample), + stream_info->channels, error)) { + g_warning("%s", error.GetMessage()); + data->unsupported = true; + return; + } + + data->frame_size = data->audio_format.GetFrameSize(); + + if (data->total_frames == 0) + data->total_frames = stream_info->total_samples; + + data->initialized = true; +} + +void flac_metadata_common_cb(const FLAC__StreamMetadata * block, + struct flac_data *data) +{ + if (data->unsupported) + return; + + struct replay_gain_info rgi; + char *mixramp_start; + char *mixramp_end; + + switch (block->type) { + case FLAC__METADATA_TYPE_STREAMINFO: + flac_got_stream_info(data, &block->data.stream_info); + break; + + case FLAC__METADATA_TYPE_VORBIS_COMMENT: + if (flac_parse_replay_gain(&rgi, block)) + decoder_replay_gain(data->decoder, &rgi); + + if (flac_parse_mixramp(&mixramp_start, &mixramp_end, block)) + decoder_mixramp(data->decoder, + mixramp_start, mixramp_end); + + flac_vorbis_comments_to_tag(data->tag, + &block->data.vorbis_comment); + + default: + break; + } +} + +/** + * This function attempts to call decoder_initialized() in case there + * was no STREAMINFO block. This is allowed for nonseekable streams, + * where the server sends us only a part of the file, without + * providing the STREAMINFO block from the beginning of the file + * (e.g. when seeking with SqueezeBox Server). + */ +static bool +flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header) +{ + if (data->unsupported) + return false; + + Error error; + if (!audio_format_init_checked(data->audio_format, + header->sample_rate, + flac_sample_format(header->bits_per_sample), + header->channels, error)) { + g_warning("%s", error.GetMessage()); + data->unsupported = true; + return false; + } + + data->frame_size = data->audio_format.GetFrameSize(); + + decoder_initialized(data->decoder, data->audio_format, + data->input_stream->seekable, + (float)data->total_frames / + (float)data->audio_format.sample_rate); + + data->initialized = true; + + return true; +} + +FLAC__StreamDecoderWriteStatus +flac_common_write(struct flac_data *data, const FLAC__Frame * frame, + const FLAC__int32 *const buf[], + FLAC__uint64 nbytes) +{ + void *buffer; + unsigned bit_rate; + + if (!data->initialized && !flac_got_first_frame(data, &frame->header)) + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; + + size_t buffer_size = frame->header.blocksize * data->frame_size; + buffer = data->buffer.Get(buffer_size); + + flac_convert(buffer, frame->header.channels, + data->audio_format.format, buf, + 0, frame->header.blocksize); + + if (nbytes > 0) + bit_rate = nbytes * 8 * frame->header.sample_rate / + (1000 * frame->header.blocksize); + else + bit_rate = 0; + + auto cmd = decoder_data(data->decoder, data->input_stream, + buffer, buffer_size, + bit_rate); + data->next_frame += frame->header.blocksize; + switch (cmd) { + case DecoderCommand::NONE: + case DecoderCommand::START: + break; + + case DecoderCommand::STOP: + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; + + case DecoderCommand::SEEK: + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; + } + + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; +} diff --git a/src/decoder/FlacCommon.hxx b/src/decoder/FlacCommon.hxx new file mode 100644 index 000000000..f9fade6fc --- /dev/null +++ b/src/decoder/FlacCommon.hxx @@ -0,0 +1,97 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* + * Common data structures and functions used by FLAC and OggFLAC + */ + +#ifndef MPD_FLAC_COMMON_HXX +#define MPD_FLAC_COMMON_HXX + +#include "FlacInput.hxx" +#include "DecoderAPI.hxx" +#include "pcm/PcmBuffer.hxx" + +#include <FLAC/stream_decoder.h> +#include <FLAC/metadata.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "flac" + +struct flac_data : public FlacInput { + PcmBuffer buffer; + + /** + * The size of one frame in the output buffer. + */ + unsigned frame_size; + + /** + * Has decoder_initialized() been called yet? + */ + bool initialized; + + /** + * Does the FLAC file contain an unsupported audio format? + */ + bool unsupported; + + /** + * The validated audio format of the FLAC file. This + * attribute is defined if "initialized" is true. + */ + AudioFormat audio_format; + + /** + * The total number of frames in this song. The decoder + * plugin may initialize this attribute to override the value + * provided by libFLAC (e.g. for sub songs from a CUE sheet). + */ + FLAC__uint64 total_frames; + + /** + * The number of the first frame in this song. This is only + * non-zero if playing sub songs from a CUE sheet. + */ + FLAC__uint64 first_frame; + + /** + * The number of the next frame which is going to be decoded. + */ + FLAC__uint64 next_frame; + + FLAC__uint64 position; + + struct decoder *decoder; + struct input_stream *input_stream; + + Tag tag; + + flac_data(struct decoder *decoder, struct input_stream *input_stream); +}; + +void flac_metadata_common_cb(const FLAC__StreamMetadata * block, + struct flac_data *data); + +FLAC__StreamDecoderWriteStatus +flac_common_write(struct flac_data *data, const FLAC__Frame * frame, + const FLAC__int32 *const buf[], + FLAC__uint64 nbytes); + +#endif /* _FLAC_COMMON_H */ diff --git a/src/decoder/FlacDecoderPlugin.cxx b/src/decoder/FlacDecoderPlugin.cxx new file mode 100644 index 000000000..a6b10fbe2 --- /dev/null +++ b/src/decoder/FlacDecoderPlugin.cxx @@ -0,0 +1,382 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" /* must be first for large file support */ +#include "FlacDecoderPlugin.h" +#include "FlacCommon.hxx" +#include "FlacMetadata.hxx" +#include "OggCodec.hxx" +#include "util/Error.hxx" + +#include <glib.h> + +#include <assert.h> +#include <unistd.h> + +#include <sys/stat.h> +#include <sys/types.h> + +#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 +#error libFLAC is too old +#endif + +static void flacPrintErroredState(FLAC__StreamDecoderState state) +{ + switch (state) { + case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA: + case FLAC__STREAM_DECODER_READ_METADATA: + case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC: + case FLAC__STREAM_DECODER_READ_FRAME: + case FLAC__STREAM_DECODER_END_OF_STREAM: + return; + + case FLAC__STREAM_DECODER_OGG_ERROR: + case FLAC__STREAM_DECODER_SEEK_ERROR: + case FLAC__STREAM_DECODER_ABORTED: + case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR: + case FLAC__STREAM_DECODER_UNINITIALIZED: + break; + } + + g_warning("%s\n", FLAC__StreamDecoderStateString[state]); +} + +static void flacMetadata(gcc_unused const FLAC__StreamDecoder * dec, + const FLAC__StreamMetadata * block, void *vdata) +{ + flac_metadata_common_cb(block, (struct flac_data *) vdata); +} + +static FLAC__StreamDecoderWriteStatus +flac_write_cb(const FLAC__StreamDecoder *dec, const FLAC__Frame *frame, + const FLAC__int32 *const buf[], void *vdata) +{ + struct flac_data *data = (struct flac_data *) vdata; + FLAC__uint64 nbytes = 0; + + if (FLAC__stream_decoder_get_decode_position(dec, &nbytes)) { + if (data->position > 0 && nbytes > data->position) { + nbytes -= data->position; + data->position += nbytes; + } else { + data->position = nbytes; + nbytes = 0; + } + } else + nbytes = 0; + + return flac_common_write(data, frame, buf, nbytes); +} + +static bool +flac_scan_file(const char *file, + const struct tag_handler *handler, void *handler_ctx) +{ + FlacMetadataChain chain; + if (!chain.Read(file)) { + g_debug("Failed to read FLAC tags: %s", + chain.GetStatusString()); + return false; + } + + chain.Scan(handler, handler_ctx); + return true; +} + +static bool +flac_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + FlacMetadataChain chain; + if (!chain.Read(is)) { + g_debug("Failed to read FLAC tags: %s", + chain.GetStatusString()); + return false; + } + + chain.Scan(handler, handler_ctx); + return true; +} + +/** + * Some glue code around FLAC__stream_decoder_new(). + */ +static FLAC__StreamDecoder * +flac_decoder_new(void) +{ + FLAC__StreamDecoder *sd = FLAC__stream_decoder_new(); + if (sd == nullptr) { + g_warning("FLAC__stream_decoder_new() failed"); + return nullptr; + } + + if(!FLAC__stream_decoder_set_metadata_respond(sd, FLAC__METADATA_TYPE_VORBIS_COMMENT)) + g_debug("FLAC__stream_decoder_set_metadata_respond() has failed"); + + return sd; +} + +static bool +flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd, + FLAC__uint64 duration) +{ + data->total_frames = duration; + + if (!FLAC__stream_decoder_process_until_end_of_metadata(sd)) { + g_warning("problem reading metadata"); + return false; + } + + if (data->initialized) { + /* done */ + decoder_initialized(data->decoder, data->audio_format, + data->input_stream->seekable, + (float)data->total_frames / + (float)data->audio_format.sample_rate); + return true; + } + + if (data->input_stream->seekable) + /* allow the workaround below only for nonseekable + streams*/ + return false; + + /* no stream_info packet found; try to initialize the decoder + from the first frame header */ + FLAC__stream_decoder_process_single(sd); + return data->initialized; +} + +static void +flac_decoder_loop(struct flac_data *data, FLAC__StreamDecoder *flac_dec, + FLAC__uint64 t_start, FLAC__uint64 t_end) +{ + struct decoder *decoder = data->decoder; + + data->first_frame = t_start; + + while (true) { + DecoderCommand cmd; + if (!data->tag.IsEmpty()) { + cmd = decoder_tag(data->decoder, data->input_stream, + std::move(data->tag)); + data->tag.Clear(); + } else + cmd = decoder_get_command(decoder); + + if (cmd == DecoderCommand::SEEK) { + FLAC__uint64 seek_sample = t_start + + decoder_seek_where(decoder) * + data->audio_format.sample_rate; + if (seek_sample >= t_start && + (t_end == 0 || seek_sample <= t_end) && + FLAC__stream_decoder_seek_absolute(flac_dec, seek_sample)) { + data->next_frame = seek_sample; + data->position = 0; + decoder_command_finished(decoder); + } else + decoder_seek_error(decoder); + } else if (cmd == DecoderCommand::STOP || + FLAC__stream_decoder_get_state(flac_dec) == FLAC__STREAM_DECODER_END_OF_STREAM) + break; + + if (t_end != 0 && data->next_frame >= t_end) + /* end of this sub track */ + break; + + if (!FLAC__stream_decoder_process_single(flac_dec) && + decoder_get_command(decoder) == DecoderCommand::NONE) { + /* a failure that was not triggered by a + decoder command */ + flacPrintErroredState(FLAC__stream_decoder_get_state(flac_dec)); + break; + } + } +} + +static FLAC__StreamDecoderInitStatus +stream_init_oggflac(FLAC__StreamDecoder *flac_dec, struct flac_data *data) +{ + return FLAC__stream_decoder_init_ogg_stream(flac_dec, + FlacInput::Read, + FlacInput::Seek, + FlacInput::Tell, + FlacInput::Length, + FlacInput::Eof, + flac_write_cb, + flacMetadata, + FlacInput::Error, + data); +} + +static FLAC__StreamDecoderInitStatus +stream_init_flac(FLAC__StreamDecoder *flac_dec, struct flac_data *data) +{ + return FLAC__stream_decoder_init_stream(flac_dec, + FlacInput::Read, + FlacInput::Seek, + FlacInput::Tell, + FlacInput::Length, + FlacInput::Eof, + flac_write_cb, + flacMetadata, + FlacInput::Error, + data); +} + +static FLAC__StreamDecoderInitStatus +stream_init(FLAC__StreamDecoder *flac_dec, struct flac_data *data, bool is_ogg) +{ + return is_ogg + ? stream_init_oggflac(flac_dec, data) + : stream_init_flac(flac_dec, data); +} + +static void +flac_decode_internal(struct decoder * decoder, + struct input_stream *input_stream, + bool is_ogg) +{ + FLAC__StreamDecoder *flac_dec; + + flac_dec = flac_decoder_new(); + if (flac_dec == nullptr) + return; + + struct flac_data data(decoder, input_stream); + + FLAC__StreamDecoderInitStatus status = + stream_init(flac_dec, &data, is_ogg); + if (status != FLAC__STREAM_DECODER_INIT_STATUS_OK) { + FLAC__stream_decoder_delete(flac_dec); + g_warning("%s", FLAC__StreamDecoderInitStatusString[status]); + return; + } + + if (!flac_decoder_initialize(&data, flac_dec, 0)) { + FLAC__stream_decoder_finish(flac_dec); + FLAC__stream_decoder_delete(flac_dec); + return; + } + + flac_decoder_loop(&data, flac_dec, 0, 0); + + FLAC__stream_decoder_finish(flac_dec); + FLAC__stream_decoder_delete(flac_dec); +} + +static void +flac_decode(struct decoder * decoder, struct input_stream *input_stream) +{ + flac_decode_internal(decoder, input_stream, false); +} + +static bool +oggflac_init(gcc_unused const config_param ¶m) +{ + return !!FLAC_API_SUPPORTS_OGG_FLAC; +} + +static bool +oggflac_scan_file(const char *file, + const struct tag_handler *handler, void *handler_ctx) +{ + FlacMetadataChain chain; + if (!chain.ReadOgg(file)) { + g_debug("Failed to read OggFLAC tags: %s", + chain.GetStatusString()); + return false; + } + + chain.Scan(handler, handler_ctx); + return true; +} + +static bool +oggflac_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + FlacMetadataChain chain; + if (!chain.ReadOgg(is)) { + g_debug("Failed to read OggFLAC tags: %s", + chain.GetStatusString()); + return false; + } + + chain.Scan(handler, handler_ctx); + return true; +} + +static void +oggflac_decode(struct decoder *decoder, struct input_stream *input_stream) +{ + if (ogg_codec_detect(decoder, input_stream) != OGG_CODEC_FLAC) + return; + + /* rewind the stream, because ogg_codec_detect() has + moved it */ + input_stream->LockSeek(0, SEEK_SET, IgnoreError()); + + flac_decode_internal(decoder, input_stream, true); +} + +static const char *const oggflac_suffixes[] = { "ogg", "oga", nullptr }; +static const char *const oggflac_mime_types[] = { + "application/ogg", + "application/x-ogg", + "audio/ogg", + "audio/x-flac+ogg", + "audio/x-ogg", + nullptr +}; + +const struct decoder_plugin oggflac_decoder_plugin = { + "oggflac", + oggflac_init, + nullptr, + oggflac_decode, + nullptr, + oggflac_scan_file, + oggflac_scan_stream, + nullptr, + oggflac_suffixes, + oggflac_mime_types, +}; + +static const char *const flac_suffixes[] = { "flac", nullptr }; +static const char *const flac_mime_types[] = { + "application/flac", + "application/x-flac", + "audio/flac", + "audio/x-flac", + nullptr +}; + +const struct decoder_plugin flac_decoder_plugin = { + "flac", + nullptr, + nullptr, + flac_decode, + nullptr, + flac_scan_file, + flac_scan_stream, + nullptr, + flac_suffixes, + flac_mime_types, +}; diff --git a/src/decoder/FlacDecoderPlugin.h b/src/decoder/FlacDecoderPlugin.h new file mode 100644 index 000000000..c99deeef7 --- /dev/null +++ b/src/decoder/FlacDecoderPlugin.h @@ -0,0 +1,26 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_FLAC_H +#define MPD_DECODER_FLAC_H + +extern const struct decoder_plugin flac_decoder_plugin; +extern const struct decoder_plugin oggflac_decoder_plugin; + +#endif diff --git a/src/decoder/FlacIOHandle.cxx b/src/decoder/FlacIOHandle.cxx new file mode 100644 index 000000000..77da864e5 --- /dev/null +++ b/src/decoder/FlacIOHandle.cxx @@ -0,0 +1,114 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "FlacIOHandle.hxx" +#include "util/Error.hxx" +#include "gcc.h" + +#include <errno.h> + +static size_t +FlacIORead(void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle) +{ + input_stream *is = (input_stream *)handle; + + uint8_t *const p0 = (uint8_t *)ptr, *p = p0, + *const end = p0 + size * nmemb; + + /* libFLAC is very picky about short reads, and expects the IO + callback to fill the whole buffer (undocumented!) */ + + Error error; + while (p < end) { + size_t nbytes = is->LockRead(p, end - p, error); + if (nbytes == 0) { + if (!error.IsDefined()) + /* end of file */ + break; + + if (error.IsDomain(errno_domain)) + errno = error.GetCode(); + else + /* just some random non-zero + errno value */ + errno = EINVAL; + return 0; + } + + p += nbytes; + } + + /* libFLAC expects a clean errno after returning from the IO + callbacks (undocumented!) */ + errno = 0; + return (p - p0) / size; +} + +static int +FlacIOSeek(FLAC__IOHandle handle, FLAC__int64 offset, int whence) +{ + input_stream *is = (input_stream *)handle; + + Error error; + return is->LockSeek(offset, whence, error) ? 0 : -1; +} + +static FLAC__int64 +FlacIOTell(FLAC__IOHandle handle) +{ + input_stream *is = (input_stream *)handle; + + return is->offset; +} + +static int +FlacIOEof(FLAC__IOHandle handle) +{ + input_stream *is = (input_stream *)handle; + + return is->LockIsEOF(); +} + +static int +FlacIOClose(gcc_unused FLAC__IOHandle handle) +{ + /* no-op because the libFLAC caller is repsonsible for closing + the #input_stream */ + + return 0; +} + +const FLAC__IOCallbacks flac_io_callbacks = { + FlacIORead, + nullptr, + nullptr, + nullptr, + FlacIOEof, + FlacIOClose, +}; + +const FLAC__IOCallbacks flac_io_callbacks_seekable = { + FlacIORead, + nullptr, + FlacIOSeek, + FlacIOTell, + FlacIOEof, + FlacIOClose, +}; diff --git a/src/decoder/FlacIOHandle.hxx b/src/decoder/FlacIOHandle.hxx new file mode 100644 index 000000000..3216dafa4 --- /dev/null +++ b/src/decoder/FlacIOHandle.hxx @@ -0,0 +1,45 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_FLAC_IO_HANDLE_HXX +#define MPD_FLAC_IO_HANDLE_HXX + +#include "gcc.h" +#include "InputStream.hxx" + +#include <FLAC/callback.h> + +extern const FLAC__IOCallbacks flac_io_callbacks; +extern const FLAC__IOCallbacks flac_io_callbacks_seekable; + +static inline FLAC__IOHandle +ToFlacIOHandle(input_stream *is) +{ + return (FLAC__IOHandle)is; +} + +static inline const FLAC__IOCallbacks & +GetFlacIOCallbacks(const input_stream *is) +{ + return is->seekable + ? flac_io_callbacks_seekable + : flac_io_callbacks; +} + +#endif diff --git a/src/decoder/FlacInput.cxx b/src/decoder/FlacInput.cxx new file mode 100644 index 000000000..19abfca81 --- /dev/null +++ b/src/decoder/FlacInput.cxx @@ -0,0 +1,149 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "FlacInput.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "util/Error.hxx" +#include "gcc.h" + +FLAC__StreamDecoderReadStatus +FlacInput::Read(FLAC__byte buffer[], size_t *bytes) +{ + size_t r = decoder_read(decoder, input_stream, (void *)buffer, *bytes); + *bytes = r; + + if (r == 0) { + if (input_stream->LockIsEOF() || + (decoder != nullptr && + decoder_get_command(decoder) != DecoderCommand::NONE)) + return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM; + else + return FLAC__STREAM_DECODER_READ_STATUS_ABORT; + } + + return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; +} + +FLAC__StreamDecoderSeekStatus +FlacInput::Seek(FLAC__uint64 absolute_byte_offset) +{ + if (!input_stream->seekable) + return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED; + + ::Error error; + if (!input_stream->LockSeek(absolute_byte_offset, SEEK_SET, error)) + return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR; + + return FLAC__STREAM_DECODER_SEEK_STATUS_OK; +} + +FLAC__StreamDecoderTellStatus +FlacInput::Tell(FLAC__uint64 *absolute_byte_offset) +{ + if (!input_stream->seekable) + return FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED; + + *absolute_byte_offset = (FLAC__uint64)input_stream->offset; + return FLAC__STREAM_DECODER_TELL_STATUS_OK; +} + +FLAC__StreamDecoderLengthStatus +FlacInput::Length(FLAC__uint64 *stream_length) +{ + if (input_stream->size < 0) + return FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED; + + *stream_length = (FLAC__uint64)input_stream->size; + return FLAC__STREAM_DECODER_LENGTH_STATUS_OK; +} + +FLAC__bool +FlacInput::Eof() +{ + return (decoder != nullptr && + decoder_get_command(decoder) != DecoderCommand::NONE && + decoder_get_command(decoder) != DecoderCommand::SEEK) || + input_stream->LockIsEOF(); +} + +void +FlacInput::Error(FLAC__StreamDecoderErrorStatus status) +{ + if (decoder == nullptr || + decoder_get_command(decoder) != DecoderCommand::STOP) + g_warning("%s", FLAC__StreamDecoderErrorStatusString[status]); +} + +FLAC__StreamDecoderReadStatus +FlacInput::Read(gcc_unused const FLAC__StreamDecoder *flac_decoder, + FLAC__byte buffer[], size_t *bytes, + void *client_data) +{ + FlacInput *i = (FlacInput *)client_data; + + return i->Read(buffer, bytes); +} + +FLAC__StreamDecoderSeekStatus +FlacInput::Seek(gcc_unused const FLAC__StreamDecoder *flac_decoder, + FLAC__uint64 absolute_byte_offset, void *client_data) +{ + FlacInput *i = (FlacInput *)client_data; + + return i->Seek(absolute_byte_offset); +} + +FLAC__StreamDecoderTellStatus +FlacInput::Tell(gcc_unused const FLAC__StreamDecoder *flac_decoder, + FLAC__uint64 *absolute_byte_offset, void *client_data) +{ + FlacInput *i = (FlacInput *)client_data; + + return i->Tell(absolute_byte_offset); +} + +FLAC__StreamDecoderLengthStatus +FlacInput::Length(gcc_unused const FLAC__StreamDecoder *flac_decoder, + FLAC__uint64 *stream_length, void *client_data) +{ + FlacInput *i = (FlacInput *)client_data; + + return i->Length(stream_length); +} + +FLAC__bool +FlacInput::Eof(gcc_unused const FLAC__StreamDecoder *flac_decoder, + void *client_data) +{ + FlacInput *i = (FlacInput *)client_data; + + return i->Eof(); +} + +void +FlacInput::Error(gcc_unused const FLAC__StreamDecoder *decoder, + FLAC__StreamDecoderErrorStatus status, void *client_data) +{ + FlacInput *i = (FlacInput *)client_data; + + i->Error(status); +} + diff --git a/src/decoder/FlacInput.hxx b/src/decoder/FlacInput.hxx new file mode 100644 index 000000000..8fc69f960 --- /dev/null +++ b/src/decoder/FlacInput.hxx @@ -0,0 +1,72 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_FLAC_INPUT_HXX +#define MPD_FLAC_INPUT_HXX + +#include <FLAC/stream_decoder.h> + +/** + * This class wraps an #input_stream in libFLAC stream decoder + * callbacks. + */ +class FlacInput { + struct decoder *decoder; + + struct input_stream *input_stream; + +public: + FlacInput(struct input_stream *_input_stream, + struct decoder *_decoder=nullptr) + :decoder(_decoder), input_stream(_input_stream) {} + +protected: + FLAC__StreamDecoderReadStatus Read(FLAC__byte buffer[], size_t *bytes); + FLAC__StreamDecoderSeekStatus Seek(FLAC__uint64 absolute_byte_offset); + FLAC__StreamDecoderTellStatus Tell(FLAC__uint64 *absolute_byte_offset); + FLAC__StreamDecoderLengthStatus Length(FLAC__uint64 *stream_length); + FLAC__bool Eof(); + void Error(FLAC__StreamDecoderErrorStatus status); + +public: + static FLAC__StreamDecoderReadStatus + Read(const FLAC__StreamDecoder *flac_decoder, + FLAC__byte buffer[], size_t *bytes, void *client_data); + + static FLAC__StreamDecoderSeekStatus + Seek(const FLAC__StreamDecoder *flac_decoder, + FLAC__uint64 absolute_byte_offset, void *client_data); + + static FLAC__StreamDecoderTellStatus + Tell(const FLAC__StreamDecoder *flac_decoder, + FLAC__uint64 *absolute_byte_offset, void *client_data); + + static FLAC__StreamDecoderLengthStatus + Length(const FLAC__StreamDecoder *flac_decoder, + FLAC__uint64 *stream_length, void *client_data); + + static FLAC__bool + Eof(const FLAC__StreamDecoder *flac_decoder, void *client_data); + + static void + Error(const FLAC__StreamDecoder *decoder, + FLAC__StreamDecoderErrorStatus status, void *client_data); +}; + +#endif diff --git a/src/decoder/FlacMetadata.cxx b/src/decoder/FlacMetadata.cxx new file mode 100644 index 000000000..49e4851e8 --- /dev/null +++ b/src/decoder/FlacMetadata.cxx @@ -0,0 +1,252 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "FlacMetadata.hxx" +#include "XiphTags.hxx" +#include "tag/Tag.hxx" +#include "tag/TagHandler.hxx" +#include "tag/TagTable.hxx" +#include "tag/TagBuilder.hxx" +#include "replay_gain_info.h" + +#include <glib.h> + +#include <assert.h> +#include <string.h> + +static bool +flac_find_float_comment(const FLAC__StreamMetadata *block, + const char *cmnt, float *fl) +{ + int offset; + size_t pos; + int len; + unsigned char tmp, *p; + + offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, + cmnt); + if (offset < 0) + return false; + + pos = strlen(cmnt) + 1; /* 1 is for '=' */ + len = block->data.vorbis_comment.comments[offset].length - pos; + if (len <= 0) + return false; + + p = &block->data.vorbis_comment.comments[offset].entry[pos]; + tmp = p[len]; + p[len] = '\0'; + *fl = (float)atof((char *)p); + p[len] = tmp; + + return true; +} + +bool +flac_parse_replay_gain(struct replay_gain_info *rgi, + const FLAC__StreamMetadata *block) +{ + bool found = false; + + replay_gain_info_init(rgi); + + if (flac_find_float_comment(block, "replaygain_album_gain", + &rgi->tuples[REPLAY_GAIN_ALBUM].gain)) + found = true; + if (flac_find_float_comment(block, "replaygain_album_peak", + &rgi->tuples[REPLAY_GAIN_ALBUM].peak)) + found = true; + if (flac_find_float_comment(block, "replaygain_track_gain", + &rgi->tuples[REPLAY_GAIN_TRACK].gain)) + found = true; + if (flac_find_float_comment(block, "replaygain_track_peak", + &rgi->tuples[REPLAY_GAIN_TRACK].peak)) + found = true; + + return found; +} + +static bool +flac_find_string_comment(const FLAC__StreamMetadata *block, + const char *cmnt, char **str) +{ + int offset; + size_t pos; + int len; + const unsigned char *p; + + *str = nullptr; + offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, + cmnt); + if (offset < 0) + return false; + + pos = strlen(cmnt) + 1; /* 1 is for '=' */ + len = block->data.vorbis_comment.comments[offset].length - pos; + if (len <= 0) + return false; + + p = &block->data.vorbis_comment.comments[offset].entry[pos]; + *str = g_strndup((const char *)p, len); + + return true; +} + +bool +flac_parse_mixramp(char **mixramp_start, char **mixramp_end, + const FLAC__StreamMetadata *block) +{ + bool found = false; + + if (flac_find_string_comment(block, "mixramp_start", mixramp_start)) + found = true; + if (flac_find_string_comment(block, "mixramp_end", mixramp_end)) + found = true; + + return found; +} + +/** + * Checks if the specified name matches the entry's name, and if yes, + * returns the comment value (not null-temrinated). + */ +static const char * +flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry, + const char *name, size_t *length_r) +{ + size_t name_length = strlen(name); + const char *comment = (const char*)entry->entry; + + if (entry->length <= name_length || + g_ascii_strncasecmp(comment, name, name_length) != 0) + return nullptr; + + if (comment[name_length] == '=') { + *length_r = entry->length - name_length - 1; + return comment + name_length + 1; + } + + return nullptr; +} + +/** + * Check if the comment's name equals the passed name, and if so, copy + * the comment value into the tag. + */ +static bool +flac_copy_comment(const FLAC__StreamMetadata_VorbisComment_Entry *entry, + const char *name, enum tag_type tag_type, + const struct tag_handler *handler, void *handler_ctx) +{ + const char *value; + size_t value_length; + + value = flac_comment_value(entry, name, &value_length); + if (value != nullptr) { + char *p = g_strndup(value, value_length); + tag_handler_invoke_tag(handler, handler_ctx, tag_type, p); + g_free(p); + return true; + } + + return false; +} + +static void +flac_scan_comment(const FLAC__StreamMetadata_VorbisComment_Entry *entry, + const struct tag_handler *handler, void *handler_ctx) +{ + if (handler->pair != nullptr) { + char *name = g_strdup((const char*)entry->entry); + char *value = strchr(name, '='); + + if (value != nullptr && value > name) { + *value++ = 0; + tag_handler_invoke_pair(handler, handler_ctx, + name, value); + } + + g_free(name); + } + + for (const struct tag_table *i = xiph_tags; i->name != nullptr; ++i) + if (flac_copy_comment(entry, i->name, i->type, + handler, handler_ctx)) + return; + + for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) + if (flac_copy_comment(entry, + tag_item_names[i], (enum tag_type)i, + handler, handler_ctx)) + return; +} + +static void +flac_scan_comments(const FLAC__StreamMetadata_VorbisComment *comment, + const struct tag_handler *handler, void *handler_ctx) +{ + for (unsigned i = 0; i < comment->num_comments; ++i) + flac_scan_comment(&comment->comments[i], + handler, handler_ctx); +} + +void +flac_scan_metadata(const FLAC__StreamMetadata *block, + const struct tag_handler *handler, void *handler_ctx) +{ + switch (block->type) { + case FLAC__METADATA_TYPE_VORBIS_COMMENT: + flac_scan_comments(&block->data.vorbis_comment, + handler, handler_ctx); + break; + + case FLAC__METADATA_TYPE_STREAMINFO: + if (block->data.stream_info.sample_rate > 0) + tag_handler_invoke_duration(handler, handler_ctx, + flac_duration(&block->data.stream_info)); + break; + + default: + break; + } +} + +void +flac_vorbis_comments_to_tag(Tag &tag, + const FLAC__StreamMetadata_VorbisComment *comment) +{ + TagBuilder tag_builder; + flac_scan_comments(comment, &add_tag_handler, &tag_builder); + tag_builder.Commit(tag); +} + +void +FlacMetadataChain::Scan(const struct tag_handler *handler, void *handler_ctx) +{ + FLACMetadataIterator iterator(*this); + + do { + FLAC__StreamMetadata *block = iterator.GetBlock(); + if (block == nullptr) + break; + + flac_scan_metadata(block, handler, handler_ctx); + } while (iterator.Next()); +} diff --git a/src/decoder/FlacMetadata.hxx b/src/decoder/FlacMetadata.hxx new file mode 100644 index 000000000..57769672f --- /dev/null +++ b/src/decoder/FlacMetadata.hxx @@ -0,0 +1,140 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_FLAC_METADATA_H +#define MPD_FLAC_METADATA_H + +#include "gcc.h" +#include "FlacIOHandle.hxx" + +#include <FLAC/metadata.h> + +#include <assert.h> + +class FlacMetadataChain { + FLAC__Metadata_Chain *chain; + +public: + FlacMetadataChain():chain(::FLAC__metadata_chain_new()) {} + + ~FlacMetadataChain() { + ::FLAC__metadata_chain_delete(chain); + } + + explicit operator FLAC__Metadata_Chain *() { + return chain; + } + + bool Read(const char *path) { + return ::FLAC__metadata_chain_read(chain, path); + } + + bool Read(FLAC__IOHandle handle, FLAC__IOCallbacks callbacks) { + return ::FLAC__metadata_chain_read_with_callbacks(chain, + handle, + callbacks); + } + + bool Read(input_stream *is) { + return Read(::ToFlacIOHandle(is), ::GetFlacIOCallbacks(is)); + } + + bool ReadOgg(const char *path) { + return ::FLAC__metadata_chain_read_ogg(chain, path); + } + + bool ReadOgg(FLAC__IOHandle handle, FLAC__IOCallbacks callbacks) { + return ::FLAC__metadata_chain_read_ogg_with_callbacks(chain, + handle, + callbacks); + } + + bool ReadOgg(input_stream *is) { + return ReadOgg(::ToFlacIOHandle(is), ::GetFlacIOCallbacks(is)); + } + + gcc_pure + FLAC__Metadata_ChainStatus GetStatus() const { + return ::FLAC__metadata_chain_status(chain); + } + + gcc_pure + const char *GetStatusString() const { + return FLAC__Metadata_ChainStatusString[GetStatus()]; + } + + void Scan(const struct tag_handler *handler, void *handler_ctx); +}; + +class FLACMetadataIterator { + FLAC__Metadata_Iterator *iterator; + +public: + FLACMetadataIterator():iterator(::FLAC__metadata_iterator_new()) {} + + FLACMetadataIterator(FlacMetadataChain &chain) + :iterator(::FLAC__metadata_iterator_new()) { + ::FLAC__metadata_iterator_init(iterator, + (FLAC__Metadata_Chain *)chain); + } + + ~FLACMetadataIterator() { + ::FLAC__metadata_iterator_delete(iterator); + } + + bool Next() { + return ::FLAC__metadata_iterator_next(iterator); + } + + gcc_pure + FLAC__StreamMetadata *GetBlock() { + return ::FLAC__metadata_iterator_get_block(iterator); + } +}; + +struct tag_handler; +struct Tag; +struct replay_gain_info; + +static inline unsigned +flac_duration(const FLAC__StreamMetadata_StreamInfo *stream_info) +{ + assert(stream_info->sample_rate > 0); + + return (stream_info->total_samples + stream_info->sample_rate - 1) / + stream_info->sample_rate; +} + +bool +flac_parse_replay_gain(struct replay_gain_info *rgi, + const FLAC__StreamMetadata *block); + +bool +flac_parse_mixramp(char **mixramp_start, char **mixramp_end, + const FLAC__StreamMetadata *block); + +void +flac_vorbis_comments_to_tag(Tag &tag, + const FLAC__StreamMetadata_VorbisComment *comment); + +void +flac_scan_metadata(const FLAC__StreamMetadata *block, + const struct tag_handler *handler, void *handler_ctx); + +#endif diff --git a/src/decoder/FlacPcm.cxx b/src/decoder/FlacPcm.cxx new file mode 100644 index 000000000..ff855fa70 --- /dev/null +++ b/src/decoder/FlacPcm.cxx @@ -0,0 +1,110 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "FlacPcm.hxx" + +#include <assert.h> + +static void flac_convert_stereo16(int16_t *dest, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + for (; position < end; ++position) { + *dest++ = buf[0][position]; + *dest++ = buf[1][position]; + } +} + +static void +flac_convert_16(int16_t *dest, + unsigned int num_channels, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan; + + for (; position < end; ++position) + for (c_chan = 0; c_chan < num_channels; c_chan++) + *dest++ = buf[c_chan][position]; +} + +/** + * Note: this function also handles 24 bit files! + */ +static void +flac_convert_32(int32_t *dest, + unsigned int num_channels, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan; + + for (; position < end; ++position) + for (c_chan = 0; c_chan < num_channels; c_chan++) + *dest++ = buf[c_chan][position]; +} + +static void +flac_convert_8(int8_t *dest, + unsigned int num_channels, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan; + + for (; position < end; ++position) + for (c_chan = 0; c_chan < num_channels; c_chan++) + *dest++ = buf[c_chan][position]; +} + +void +flac_convert(void *dest, + unsigned int num_channels, SampleFormat sample_format, + const FLAC__int32 *const buf[], + unsigned int position, unsigned int end) +{ + switch (sample_format) { + case SampleFormat::S16: + if (num_channels == 2) + flac_convert_stereo16((int16_t*)dest, buf, + position, end); + else + flac_convert_16((int16_t*)dest, num_channels, buf, + position, end); + break; + + case SampleFormat::S24_P32: + case SampleFormat::S32: + flac_convert_32((int32_t*)dest, num_channels, buf, + position, end); + break; + + case SampleFormat::S8: + flac_convert_8((int8_t*)dest, num_channels, buf, + position, end); + break; + + case SampleFormat::FLOAT: + case SampleFormat::DSD: + case SampleFormat::UNDEFINED: + assert(false); + gcc_unreachable(); + } +} diff --git a/src/decoder/FlacPcm.hxx b/src/decoder/FlacPcm.hxx new file mode 100644 index 000000000..fa85f65dd --- /dev/null +++ b/src/decoder/FlacPcm.hxx @@ -0,0 +1,33 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_FLAC_PCM_HXX +#define MPD_FLAC_PCM_HXX + +#include "AudioFormat.hxx" + +#include <FLAC/ordinals.h> + +void +flac_convert(void *dest, + unsigned int num_channels, SampleFormat sample_format, + const FLAC__int32 *const buf[], + unsigned int position, unsigned int end); + +#endif diff --git a/src/decoder/FluidsynthDecoderPlugin.cxx b/src/decoder/FluidsynthDecoderPlugin.cxx new file mode 100644 index 000000000..4db4f1618 --- /dev/null +++ b/src/decoder/FluidsynthDecoderPlugin.cxx @@ -0,0 +1,223 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "FluidsynthDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "CheckAudioFormat.hxx" +#include "util/Error.hxx" + +#include <glib.h> + +#include <fluidsynth.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "fluidsynth" + +static unsigned sample_rate; +static const char *soundfont_path; + +/** + * Convert a fluidsynth log level to a GLib log level. + */ +static GLogLevelFlags +fluidsynth_level_to_glib(enum fluid_log_level level) +{ + switch (level) { + case FLUID_PANIC: + case FLUID_ERR: + return G_LOG_LEVEL_CRITICAL; + + case FLUID_WARN: + return G_LOG_LEVEL_WARNING; + + case FLUID_INFO: + return G_LOG_LEVEL_INFO; + + case FLUID_DBG: + case LAST_LOG_LEVEL: + return G_LOG_LEVEL_DEBUG; + } + + /* invalid fluidsynth log level */ + return G_LOG_LEVEL_MESSAGE; +} + +/** + * The fluidsynth logging callback. It forwards messages to the GLib + * logging library. + */ +static void +fluidsynth_mpd_log_function(int level, char *message, gcc_unused void *data) +{ + g_log(G_LOG_DOMAIN, fluidsynth_level_to_glib(fluid_log_level(level)), + "%s", message); +} + +static bool +fluidsynth_init(const config_param ¶m) +{ + Error error; + + sample_rate = param.GetBlockValue("sample_rate", 48000u); + if (!audio_check_sample_rate(sample_rate, error)) { + g_warning("%s", error.GetMessage()); + return false; + } + + soundfont_path = param.GetBlockValue("soundfont", + "/usr/share/sounds/sf2/FluidR3_GM.sf2"); + + fluid_set_log_function(LAST_LOG_LEVEL, + fluidsynth_mpd_log_function, nullptr); + + return true; +} + +static void +fluidsynth_file_decode(struct decoder *decoder, const char *path_fs) +{ + char setting_sample_rate[] = "synth.sample-rate"; + /* + char setting_verbose[] = "synth.verbose"; + char setting_yes[] = "yes"; + */ + fluid_settings_t *settings; + fluid_synth_t *synth; + fluid_player_t *player; + int ret; + + /* set up fluid settings */ + + settings = new_fluid_settings(); + if (settings == nullptr) + return; + + fluid_settings_setnum(settings, setting_sample_rate, sample_rate); + + /* + fluid_settings_setstr(settings, setting_verbose, setting_yes); + */ + + /* create the fluid synth */ + + synth = new_fluid_synth(settings); + if (synth == nullptr) { + delete_fluid_settings(settings); + return; + } + + ret = fluid_synth_sfload(synth, soundfont_path, true); + if (ret < 0) { + g_warning("fluid_synth_sfload() failed"); + delete_fluid_synth(synth); + delete_fluid_settings(settings); + return; + } + + /* create the fluid player */ + + player = new_fluid_player(synth); + if (player == nullptr) { + delete_fluid_synth(synth); + delete_fluid_settings(settings); + return; + } + + ret = fluid_player_add(player, path_fs); + if (ret != 0) { + g_warning("fluid_player_add() failed"); + delete_fluid_player(player); + delete_fluid_synth(synth); + delete_fluid_settings(settings); + return; + } + + /* start the player */ + + ret = fluid_player_play(player); + if (ret != 0) { + g_warning("fluid_player_play() failed"); + delete_fluid_player(player); + delete_fluid_synth(synth); + delete_fluid_settings(settings); + return; + } + + /* initialization complete - announce the audio format to the + MPD core */ + + const AudioFormat audio_format(sample_rate, SampleFormat::S16, 2); + decoder_initialized(decoder, audio_format, false, -1); + + DecoderCommand cmd; + while (fluid_player_get_status(player) == FLUID_PLAYER_PLAYING) { + int16_t buffer[2048]; + const unsigned max_frames = G_N_ELEMENTS(buffer) / 2; + + /* read samples from fluidsynth and send them to the + MPD core */ + + ret = fluid_synth_write_s16(synth, max_frames, + buffer, 0, 2, + buffer, 1, 2); + if (ret != 0) + break; + + cmd = decoder_data(decoder, nullptr, buffer, sizeof(buffer), + 0); + if (cmd != DecoderCommand::NONE) + break; + } + + /* clean up */ + + fluid_player_stop(player); + fluid_player_join(player); + + delete_fluid_player(player); + delete_fluid_synth(synth); + delete_fluid_settings(settings); +} + +static bool +fluidsynth_scan_file(const char *file, + gcc_unused const struct tag_handler *handler, + gcc_unused void *handler_ctx) +{ + return fluid_is_midifile(file); +} + +static const char *const fluidsynth_suffixes[] = { + "mid", + nullptr +}; + +const struct decoder_plugin fluidsynth_decoder_plugin = { + "fluidsynth", + fluidsynth_init, + nullptr, + nullptr, + fluidsynth_file_decode, + fluidsynth_scan_file, + nullptr, + nullptr, + fluidsynth_suffixes, + nullptr, +}; diff --git a/src/decoder/FluidsynthDecoderPlugin.hxx b/src/decoder/FluidsynthDecoderPlugin.hxx new file mode 100644 index 000000000..40ed7e4d8 --- /dev/null +++ b/src/decoder/FluidsynthDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_FLUIDSYNTH_HXX +#define MPD_DECODER_FLUIDSYNTH_HXX + +extern const struct decoder_plugin fluidsynth_decoder_plugin; + +#endif diff --git a/src/decoder/GmeDecoderPlugin.cxx b/src/decoder/GmeDecoderPlugin.cxx new file mode 100644 index 000000000..dbe1d000f --- /dev/null +++ b/src/decoder/GmeDecoderPlugin.cxx @@ -0,0 +1,289 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "GmeDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "util/UriUtil.hxx" +#include "util/Error.hxx" + +#include <glib.h> +#include <assert.h> +#include <stdlib.h> +#include <string.h> + +#include <gme/gme.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "gme" + +#define SUBTUNE_PREFIX "tune_" + +static constexpr unsigned GME_SAMPLE_RATE = 44100; +static constexpr unsigned GME_CHANNELS = 2; +static constexpr unsigned GME_BUFFER_FRAMES = 2048; +static constexpr unsigned GME_BUFFER_SAMPLES = + GME_BUFFER_FRAMES * GME_CHANNELS; + +/** + * returns the file path stripped of any /tune_xxx.* subtune + * suffix + */ +static char * +get_container_name(const char *path_fs) +{ + const char *subtune_suffix = uri_get_suffix(path_fs); + char *path_container = g_strdup(path_fs); + char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", + subtune_suffix, nullptr); + GPatternSpec *path_with_subtune = g_pattern_spec_new(pat); + g_free(pat); + if (!g_pattern_match(path_with_subtune, + strlen(path_container), path_container, nullptr)) { + g_pattern_spec_free(path_with_subtune); + return path_container; + } + + char *ptr = g_strrstr(path_container, "/" SUBTUNE_PREFIX); + if (ptr != nullptr) + *ptr='\0'; + + g_pattern_spec_free(path_with_subtune); + return path_container; +} + +/** + * returns tune number from file.nsf/tune_xxx.* style path or 0 if no subtune + * is appended. + */ +static int +get_song_num(const char *path_fs) +{ + const char *subtune_suffix = uri_get_suffix(path_fs); + char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", + subtune_suffix, nullptr); + GPatternSpec *path_with_subtune = g_pattern_spec_new(pat); + g_free(pat); + + if (g_pattern_match(path_with_subtune, + strlen(path_fs), path_fs, nullptr)) { + char *sub = g_strrstr(path_fs, "/" SUBTUNE_PREFIX); + g_pattern_spec_free(path_with_subtune); + if (!sub) + return 0; + + sub += strlen("/" SUBTUNE_PREFIX); + int song_num = strtol(sub, nullptr, 10); + + return song_num - 1; + } else { + g_pattern_spec_free(path_with_subtune); + return 0; + } +} + +static char * +gme_container_scan(const char *path_fs, const unsigned int tnum) +{ + Music_Emu *emu; + const char *gme_err = gme_open_file(path_fs, &emu, GME_SAMPLE_RATE); + if (gme_err != nullptr) { + g_warning("%s", gme_err); + return nullptr; + } + + const unsigned num_songs = gme_track_count(emu); + /* if it only contains a single tune, don't treat as container */ + if (num_songs < 2) + return nullptr; + + const char *subtune_suffix = uri_get_suffix(path_fs); + if (tnum <= num_songs){ + char *subtune = g_strdup_printf( + SUBTUNE_PREFIX "%03u.%s", tnum, subtune_suffix); + return subtune; + } else + return nullptr; +} + +static void +gme_file_decode(struct decoder *decoder, const char *path_fs) +{ + char *path_container = get_container_name(path_fs); + + Music_Emu *emu; + const char *gme_err = + gme_open_file(path_container, &emu, GME_SAMPLE_RATE); + g_free(path_container); + if (gme_err != nullptr) { + g_warning("%s", gme_err); + return; + } + + gme_info_t *ti; + const int song_num = get_song_num(path_fs); + gme_err = gme_track_info(emu, &ti, song_num); + if (gme_err != nullptr) { + g_warning("%s", gme_err); + gme_delete(emu); + return; + } + + const float song_len = ti->length > 0 + ? ti->length / 1000.0 + : -1.0; + + /* initialize the MPD decoder */ + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, GME_SAMPLE_RATE, + SampleFormat::S16, GME_CHANNELS, + error)) { + g_warning("%s", error.GetMessage()); + gme_free_info(ti); + gme_delete(emu); + return; + } + + decoder_initialized(decoder, audio_format, true, song_len); + + gme_err = gme_start_track(emu, song_num); + if (gme_err != nullptr) + g_warning("%s", gme_err); + + if (ti->length > 0) + gme_set_fade(emu, ti->length); + + /* play */ + DecoderCommand cmd; + do { + short buf[GME_BUFFER_SAMPLES]; + gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf); + if (gme_err != nullptr) { + g_warning("%s", gme_err); + return; + } + + cmd = decoder_data(decoder, nullptr, buf, sizeof(buf), 0); + if (cmd == DecoderCommand::SEEK) { + float where = decoder_seek_where(decoder); + gme_err = gme_seek(emu, int(where * 1000)); + if (gme_err != nullptr) + g_warning("%s", gme_err); + decoder_command_finished(decoder); + } + + if (gme_track_ended(emu)) + break; + } while (cmd != DecoderCommand::STOP); + + gme_free_info(ti); + gme_delete(emu); +} + +static bool +gme_scan_file(const char *path_fs, + const struct tag_handler *handler, void *handler_ctx) +{ + char *path_container = get_container_name(path_fs); + + Music_Emu *emu; + const char *gme_err = + gme_open_file(path_container, &emu, GME_SAMPLE_RATE); + g_free(path_container); + if (gme_err != nullptr) { + g_warning("%s", gme_err); + return false; + } + + const int song_num = get_song_num(path_fs); + + gme_info_t *ti; + gme_err = gme_track_info(emu, &ti, song_num); + if (gme_err != nullptr) { + g_warning("%s", gme_err); + gme_delete(emu); + return false; + } + + assert(ti != nullptr); + + if (ti->length > 0) + tag_handler_invoke_duration(handler, handler_ctx, + ti->length / 100); + + if (ti->song != nullptr) { + if (gme_track_count(emu) > 1) { + /* start numbering subtunes from 1 */ + char *tag_title = + g_strdup_printf("%s (%d/%d)", + ti->song, song_num + 1, + gme_track_count(emu)); + tag_handler_invoke_tag(handler, handler_ctx, + TAG_TITLE, tag_title); + g_free(tag_title); + } else + tag_handler_invoke_tag(handler, handler_ctx, + TAG_TITLE, ti->song); + } + + if (ti->author != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_ARTIST, ti->author); + + if (ti->game != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_ALBUM, ti->game); + + if (ti->comment != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_COMMENT, ti->comment); + + if (ti->copyright != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_DATE, ti->copyright); + + gme_free_info(ti); + gme_delete(emu); + + return true; +} + +static const char *const gme_suffixes[] = { + "ay", "gbs", "gym", "hes", "kss", "nsf", + "nsfe", "sap", "spc", "vgm", "vgz", + nullptr +}; + +extern const struct decoder_plugin gme_decoder_plugin; +const struct decoder_plugin gme_decoder_plugin = { + "gme", + nullptr, + nullptr, + nullptr, + gme_file_decode, + gme_scan_file, + nullptr, + gme_container_scan, + gme_suffixes, + nullptr, +}; diff --git a/src/decoder/GmeDecoderPlugin.hxx b/src/decoder/GmeDecoderPlugin.hxx new file mode 100644 index 000000000..fba735d92 --- /dev/null +++ b/src/decoder/GmeDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_GME_HXX +#define MPD_DECODER_GME_HXX + +extern const struct decoder_plugin gme_decoder_plugin; + +#endif diff --git a/src/decoder/MadDecoderPlugin.cxx b/src/decoder/MadDecoderPlugin.cxx new file mode 100644 index 000000000..b7d90892b --- /dev/null +++ b/src/decoder/MadDecoderPlugin.cxx @@ -0,0 +1,1180 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "MadDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "ConfigGlobal.hxx" +#include "tag/TagId3.hxx" +#include "tag/TagRva2.hxx" +#include "tag/TagHandler.hxx" +#include "CheckAudioFormat.hxx" +#include "util/Error.hxx" + +#include <assert.h> +#include <unistd.h> +#include <stdlib.h> +#include <stdio.h> +#include <string.h> +#include <glib.h> +#include <mad.h> + +#ifdef HAVE_ID3TAG +#include <id3tag.h> +#endif + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "mad" + +#define FRAMES_CUSHION 2000 + +#define READ_BUFFER_SIZE 40960 + +enum mp3_action { + DECODE_SKIP = -3, + DECODE_BREAK = -2, + DECODE_CONT = -1, + DECODE_OK = 0 +}; + +enum muteframe { + MUTEFRAME_NONE, + MUTEFRAME_SKIP, + MUTEFRAME_SEEK +}; + +/* the number of samples of silence the decoder inserts at start */ +#define DECODERDELAY 529 + +#define DEFAULT_GAPLESS_MP3_PLAYBACK true + +static bool gapless_playback; + +static inline int32_t +mad_fixed_to_24_sample(mad_fixed_t sample) +{ + enum { + bits = 24, + MIN = -MAD_F_ONE, + MAX = MAD_F_ONE - 1 + }; + + /* round */ + sample = sample + (1L << (MAD_F_FRACBITS - bits)); + + /* clip */ + if (gcc_unlikely(sample > MAX)) + sample = MAX; + else if (gcc_unlikely(sample < MIN)) + sample = MIN; + + /* quantize */ + return sample >> (MAD_F_FRACBITS + 1 - bits); +} + +static void +mad_fixed_to_24_buffer(int32_t *dest, const struct mad_synth *synth, + unsigned int start, unsigned int end, + unsigned int num_channels) +{ + unsigned int i, c; + + for (i = start; i < end; ++i) { + for (c = 0; c < num_channels; ++c) + *dest++ = mad_fixed_to_24_sample(synth->pcm.samples[c][i]); + } +} + +static bool +mp3_plugin_init(gcc_unused const config_param ¶m) +{ + gapless_playback = config_get_bool(CONF_GAPLESS_MP3_PLAYBACK, + DEFAULT_GAPLESS_MP3_PLAYBACK); + return true; +} + +#define MP3_DATA_OUTPUT_BUFFER_SIZE 2048 + +struct MadDecoder { + struct mad_stream stream; + struct mad_frame frame; + struct mad_synth synth; + mad_timer_t timer; + unsigned char input_buffer[READ_BUFFER_SIZE]; + int32_t output_buffer[MP3_DATA_OUTPUT_BUFFER_SIZE]; + float total_time; + float elapsed_time; + float seek_where; + enum muteframe mute_frame; + long *frame_offsets; + mad_timer_t *times; + unsigned long highest_frame; + unsigned long max_frames; + unsigned long current_frame; + unsigned int drop_start_frames; + unsigned int drop_end_frames; + unsigned int drop_start_samples; + unsigned int drop_end_samples; + bool found_replay_gain; + bool found_xing; + bool found_first_frame; + bool decoded_first_frame; + unsigned long bit_rate; + struct decoder *decoder; + struct input_stream *input_stream; + enum mad_layer layer; + + MadDecoder(struct decoder *decoder, struct input_stream *input_stream); + ~MadDecoder(); + + bool Seek(long offset); + bool FillBuffer(); + void ParseId3(size_t tagsize, Tag **mpd_tag); + enum mp3_action DecodeNextFrameHeader(Tag **tag); + enum mp3_action DecodeNextFrame(); + + gcc_pure + goffset ThisFrameOffset() const; + + gcc_pure + goffset RestIncludingThisFrame() const; + + /** + * Attempt to calulcate the length of the song from filesize + */ + void FileSizeToSongLength(); + + bool DecodeFirstFrame(Tag **tag); + + gcc_pure + long TimeToFrame(double t) const; + + void UpdateTimerNextFrame(); + + /** + * Sends the synthesized current frame via decoder_data(). + */ + DecoderCommand SendPCM(unsigned i, unsigned pcm_length); + + /** + * Synthesize the current frame and send it via + * decoder_data(). + */ + DecoderCommand SyncAndSend(); + + bool Read(); +}; + +MadDecoder::MadDecoder(struct decoder *_decoder, + struct input_stream *_input_stream) + :mute_frame(MUTEFRAME_NONE), + frame_offsets(nullptr), + times(nullptr), + highest_frame(0), max_frames(0), current_frame(0), + drop_start_frames(0), drop_end_frames(0), + drop_start_samples(0), drop_end_samples(0), + found_replay_gain(false), found_xing(false), + found_first_frame(false), decoded_first_frame(false), + decoder(_decoder), input_stream(_input_stream), + layer(mad_layer(0)) +{ + mad_stream_init(&stream); + mad_stream_options(&stream, MAD_OPTION_IGNORECRC); + mad_frame_init(&frame); + mad_synth_init(&synth); + mad_timer_reset(&timer); +} + +inline bool +MadDecoder::Seek(long offset) +{ + Error error; + if (!input_stream->LockSeek(offset, SEEK_SET, error)) + return false; + + mad_stream_buffer(&stream, input_buffer, 0); + stream.error = MAD_ERROR_NONE; + + return true; +} + +inline bool +MadDecoder::FillBuffer() +{ + size_t remaining, length; + unsigned char *dest; + + if (stream.next_frame != nullptr) { + remaining = stream.bufend - stream.next_frame; + memmove(input_buffer, stream.next_frame, remaining); + dest = input_buffer + remaining; + length = READ_BUFFER_SIZE - remaining; + } else { + remaining = 0; + length = READ_BUFFER_SIZE; + dest = input_buffer; + } + + /* we've exhausted the read buffer, so give up!, these potential + * mp3 frames are way too big, and thus unlikely to be mp3 frames */ + if (length == 0) + return false; + + length = decoder_read(decoder, input_stream, dest, length); + if (length == 0) + return false; + + mad_stream_buffer(&stream, input_buffer, length + remaining); + stream.error = MAD_ERROR_NONE; + + return true; +} + +#ifdef HAVE_ID3TAG +static bool +parse_id3_replay_gain_info(struct replay_gain_info *replay_gain_info, + struct id3_tag *tag) +{ + int i; + char *key; + char *value; + struct id3_frame *frame; + bool found = false; + + replay_gain_info_init(replay_gain_info); + + for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) { + if (frame->nfields < 3) + continue; + + key = (char *) + id3_ucs4_latin1duplicate(id3_field_getstring + (&frame->fields[1])); + value = (char *) + id3_ucs4_latin1duplicate(id3_field_getstring + (&frame->fields[2])); + + if (g_ascii_strcasecmp(key, "replaygain_track_gain") == 0) { + replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = atof(value); + found = true; + } else if (g_ascii_strcasecmp(key, "replaygain_album_gain") == 0) { + replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = atof(value); + found = true; + } else if (g_ascii_strcasecmp(key, "replaygain_track_peak") == 0) { + replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = atof(value); + found = true; + } else if (g_ascii_strcasecmp(key, "replaygain_album_peak") == 0) { + replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = atof(value); + found = true; + } + + free(key); + free(value); + } + + return found || + /* fall back on RVA2 if no replaygain tags found */ + tag_rva2_parse(tag, replay_gain_info); +} +#endif + +#ifdef HAVE_ID3TAG +static bool +parse_id3_mixramp(char **mixramp_start, char **mixramp_end, + struct id3_tag *tag) +{ + int i; + char *key; + char *value; + struct id3_frame *frame; + bool found = false; + + *mixramp_start = nullptr; + *mixramp_end = nullptr; + + for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) { + if (frame->nfields < 3) + continue; + + key = (char *) + id3_ucs4_latin1duplicate(id3_field_getstring + (&frame->fields[1])); + value = (char *) + id3_ucs4_latin1duplicate(id3_field_getstring + (&frame->fields[2])); + + if (g_ascii_strcasecmp(key, "mixramp_start") == 0) { + *mixramp_start = g_strdup(value); + found = true; + } else if (g_ascii_strcasecmp(key, "mixramp_end") == 0) { + *mixramp_end = g_strdup(value); + found = true; + } + + free(key); + free(value); + } + + return found; +} +#endif + +inline void +MadDecoder::ParseId3(size_t tagsize, Tag **mpd_tag) +{ +#ifdef HAVE_ID3TAG + struct id3_tag *id3_tag = nullptr; + id3_length_t count; + id3_byte_t const *id3_data; + id3_byte_t *allocated = nullptr; + + count = stream.bufend - stream.this_frame; + + if (tagsize <= count) { + id3_data = stream.this_frame; + mad_stream_skip(&(stream), tagsize); + } else { + allocated = (id3_byte_t *)g_malloc(tagsize); + memcpy(allocated, stream.this_frame, count); + mad_stream_skip(&(stream), count); + + while (count < tagsize) { + size_t len; + + len = decoder_read(decoder, input_stream, + allocated + count, tagsize - count); + if (len == 0) + break; + else + count += len; + } + + if (count != tagsize) { + g_debug("error parsing ID3 tag"); + g_free(allocated); + return; + } + + id3_data = allocated; + } + + id3_tag = id3_tag_parse(id3_data, tagsize); + if (id3_tag == nullptr) { + g_free(allocated); + return; + } + + if (mpd_tag) { + Tag *tmp_tag = tag_id3_import(id3_tag); + if (tmp_tag != nullptr) { + delete *mpd_tag; + *mpd_tag = tmp_tag; + } + } + + if (decoder != nullptr) { + struct replay_gain_info rgi; + char *mixramp_start; + char *mixramp_end; + + if (parse_id3_replay_gain_info(&rgi, id3_tag)) { + decoder_replay_gain(decoder, &rgi); + found_replay_gain = true; + } + + if (parse_id3_mixramp(&mixramp_start, &mixramp_end, id3_tag)) + decoder_mixramp(decoder, mixramp_start, mixramp_end); + } + + id3_tag_delete(id3_tag); + + g_free(allocated); +#else /* !HAVE_ID3TAG */ + (void)mpd_tag; + + /* This code is enabled when libid3tag is disabled. Instead + of parsing the ID3 frame, it just skips it. */ + + size_t count = stream.bufend - stream.this_frame; + + if (tagsize <= count) { + mad_stream_skip(&stream, tagsize); + } else { + mad_stream_skip(&stream, count); + + while (count < tagsize) { + size_t len = tagsize - count; + char ignored[1024]; + if (len > sizeof(ignored)) + len = sizeof(ignored); + + len = decoder_read(decoder, input_stream, + ignored, len); + if (len == 0) + break; + else + count += len; + } + } +#endif +} + +#ifndef HAVE_ID3TAG +/** + * This function emulates libid3tag when it is disabled. Instead of + * doing a real analyzation of the frame, it just checks whether the + * frame begins with the string "ID3". If so, it returns the length + * of the ID3 frame. + */ +static signed long +id3_tag_query(const void *p0, size_t length) +{ + const char *p = (const char *)p0; + + return length >= 10 && memcmp(p, "ID3", 3) == 0 + ? (p[8] << 7) + p[9] + 10 + : 0; +} +#endif /* !HAVE_ID3TAG */ + +enum mp3_action +MadDecoder::DecodeNextFrameHeader(Tag **tag) +{ + if ((stream.buffer == nullptr || stream.error == MAD_ERROR_BUFLEN) && + !FillBuffer()) + return DECODE_BREAK; + + if (mad_header_decode(&frame.header, &stream)) { + if (stream.error == MAD_ERROR_LOSTSYNC && stream.this_frame) { + signed long tagsize = id3_tag_query(stream.this_frame, + stream.bufend - + stream.this_frame); + + if (tagsize > 0) { + if (tag && !(*tag)) { + ParseId3((size_t)tagsize, tag); + } else { + mad_stream_skip(&stream, tagsize); + } + return DECODE_CONT; + } + } + if (MAD_RECOVERABLE(stream.error)) { + return DECODE_SKIP; + } else { + if (stream.error == MAD_ERROR_BUFLEN) + return DECODE_CONT; + else { + g_warning("unrecoverable frame level error " + "(%s).\n", + mad_stream_errorstr(&stream)); + return DECODE_BREAK; + } + } + } + + enum mad_layer new_layer = frame.header.layer; + if (layer == (mad_layer)0) { + if (new_layer != MAD_LAYER_II && new_layer != MAD_LAYER_III) { + /* Only layer 2 and 3 have been tested to work */ + return DECODE_SKIP; + } + + layer = new_layer; + } else if (new_layer != layer) { + /* Don't decode frames with a different layer than the first */ + return DECODE_SKIP; + } + + return DECODE_OK; +} + +enum mp3_action +MadDecoder::DecodeNextFrame() +{ + if ((stream.buffer == nullptr || stream.error == MAD_ERROR_BUFLEN) && + !FillBuffer()) + return DECODE_BREAK; + + if (mad_frame_decode(&frame, &stream)) { + if (stream.error == MAD_ERROR_LOSTSYNC) { + signed long tagsize = id3_tag_query(stream.this_frame, + stream.bufend - + stream.this_frame); + if (tagsize > 0) { + mad_stream_skip(&stream, tagsize); + return DECODE_CONT; + } + } + if (MAD_RECOVERABLE(stream.error)) { + return DECODE_SKIP; + } else { + if (stream.error == MAD_ERROR_BUFLEN) + return DECODE_CONT; + else { + g_warning("unrecoverable frame level error " + "(%s).\n", + mad_stream_errorstr(&stream)); + return DECODE_BREAK; + } + } + } + + return DECODE_OK; +} + +/* xing stuff stolen from alsaplayer, and heavily modified by jat */ +#define XI_MAGIC (('X' << 8) | 'i') +#define NG_MAGIC (('n' << 8) | 'g') +#define IN_MAGIC (('I' << 8) | 'n') +#define FO_MAGIC (('f' << 8) | 'o') + +enum xing_magic { + XING_MAGIC_XING, /* VBR */ + XING_MAGIC_INFO /* CBR */ +}; + +struct xing { + long flags; /* valid fields (see below) */ + unsigned long frames; /* total number of frames */ + unsigned long bytes; /* total number of bytes */ + unsigned char toc[100]; /* 100-point seek table */ + long scale; /* VBR quality */ + enum xing_magic magic; /* header magic */ +}; + +enum { + XING_FRAMES = 0x00000001L, + XING_BYTES = 0x00000002L, + XING_TOC = 0x00000004L, + XING_SCALE = 0x00000008L +}; + +struct lame_version { + unsigned major; + unsigned minor; +}; + +struct lame { + char encoder[10]; /* 9 byte encoder name/version ("LAME3.97b") */ + struct lame_version version; /* struct containing just the version */ + float peak; /* replaygain peak */ + float track_gain; /* replaygain track gain */ + float album_gain; /* replaygain album gain */ + int encoder_delay; /* # of added samples at start of mp3 */ + int encoder_padding; /* # of added samples at end of mp3 */ + int crc; /* CRC of the first 190 bytes of this frame */ +}; + +static bool +parse_xing(struct xing *xing, struct mad_bitptr *ptr, int *oldbitlen) +{ + unsigned long bits; + int bitlen; + int bitsleft; + int i; + + bitlen = *oldbitlen; + + if (bitlen < 16) + return false; + + bits = mad_bit_read(ptr, 16); + bitlen -= 16; + + if (bits == XI_MAGIC) { + if (bitlen < 16) + return false; + + if (mad_bit_read(ptr, 16) != NG_MAGIC) + return false; + + bitlen -= 16; + xing->magic = XING_MAGIC_XING; + } else if (bits == IN_MAGIC) { + if (bitlen < 16) + return false; + + if (mad_bit_read(ptr, 16) != FO_MAGIC) + return false; + + bitlen -= 16; + xing->magic = XING_MAGIC_INFO; + } + else if (bits == NG_MAGIC) xing->magic = XING_MAGIC_XING; + else if (bits == FO_MAGIC) xing->magic = XING_MAGIC_INFO; + else + return false; + + if (bitlen < 32) + return false; + xing->flags = mad_bit_read(ptr, 32); + bitlen -= 32; + + if (xing->flags & XING_FRAMES) { + if (bitlen < 32) + return false; + xing->frames = mad_bit_read(ptr, 32); + bitlen -= 32; + } + + if (xing->flags & XING_BYTES) { + if (bitlen < 32) + return false; + xing->bytes = mad_bit_read(ptr, 32); + bitlen -= 32; + } + + if (xing->flags & XING_TOC) { + if (bitlen < 800) + return false; + for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(ptr, 8); + bitlen -= 800; + } + + if (xing->flags & XING_SCALE) { + if (bitlen < 32) + return false; + xing->scale = mad_bit_read(ptr, 32); + bitlen -= 32; + } + + /* Make sure we consume no less than 120 bytes (960 bits) in hopes that + * the LAME tag is found there, and not right after the Xing header */ + bitsleft = 960 - ((*oldbitlen) - bitlen); + if (bitsleft < 0) + return false; + else if (bitsleft > 0) { + mad_bit_read(ptr, bitsleft); + bitlen -= bitsleft; + } + + *oldbitlen = bitlen; + + return true; +} + +static bool +parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) +{ + int adj = 0; + int name; + int orig; + int sign; + int gain; + int i; + + /* Unlike the xing header, the lame tag has a fixed length. Fail if + * not all 36 bytes (288 bits) are there. */ + if (*bitlen < 288) + return false; + + for (i = 0; i < 9; i++) + lame->encoder[i] = (char)mad_bit_read(ptr, 8); + lame->encoder[9] = '\0'; + + *bitlen -= 72; + + /* This is technically incorrect, since the encoder might not be lame. + * But there's no other way to determine if this is a lame tag, and we + * wouldn't want to go reading a tag that's not there. */ + if (!g_str_has_prefix(lame->encoder, "LAME")) + return false; + + if (sscanf(lame->encoder+4, "%u.%u", + &lame->version.major, &lame->version.minor) != 2) + return false; + + g_debug("detected LAME version %i.%i (\"%s\")\n", + lame->version.major, lame->version.minor, lame->encoder); + + /* The reference volume was changed from the 83dB used in the + * ReplayGain spec to 89dB in lame 3.95.1. Bump the gain for older + * versions, since everyone else uses 89dB instead of 83dB. + * Unfortunately, lame didn't differentiate between 3.95 and 3.95.1, so + * it's impossible to make the proper adjustment for 3.95. + * Fortunately, 3.95 was only out for about a day before 3.95.1 was + * released. -- tmz */ + if (lame->version.major < 3 || + (lame->version.major == 3 && lame->version.minor < 95)) + adj = 6; + + mad_bit_read(ptr, 16); + + lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */ + g_debug("LAME peak found: %f\n", lame->peak); + + lame->track_gain = 0; + name = mad_bit_read(ptr, 3); /* gain name */ + orig = mad_bit_read(ptr, 3); /* gain originator */ + sign = mad_bit_read(ptr, 1); /* sign bit */ + gain = mad_bit_read(ptr, 9); /* gain*10 */ + if (gain && name == 1 && orig != 0) { + lame->track_gain = ((sign ? -gain : gain) / 10.0) + adj; + g_debug("LAME track gain found: %f\n", lame->track_gain); + } + + /* tmz reports that this isn't currently written by any version of lame + * (as of 3.97). Since we have no way of testing it, don't use it. + * Wouldn't want to go blowing someone's ears just because we read it + * wrong. :P -- jat */ + lame->album_gain = 0; +#if 0 + name = mad_bit_read(ptr, 3); /* gain name */ + orig = mad_bit_read(ptr, 3); /* gain originator */ + sign = mad_bit_read(ptr, 1); /* sign bit */ + gain = mad_bit_read(ptr, 9); /* gain*10 */ + if (gain && name == 2 && orig != 0) { + lame->album_gain = ((sign ? -gain : gain) / 10.0) + adj; + g_debug("LAME album gain found: %f\n", lame->track_gain); + } +#else + mad_bit_read(ptr, 16); +#endif + + mad_bit_read(ptr, 16); + + lame->encoder_delay = mad_bit_read(ptr, 12); + lame->encoder_padding = mad_bit_read(ptr, 12); + + g_debug("encoder delay is %i, encoder padding is %i\n", + lame->encoder_delay, lame->encoder_padding); + + mad_bit_read(ptr, 80); + + lame->crc = mad_bit_read(ptr, 16); + + *bitlen -= 216; + + return true; +} + +static inline float +mp3_frame_duration(const struct mad_frame *frame) +{ + return mad_timer_count(frame->header.duration, + MAD_UNITS_MILLISECONDS) / 1000.0; +} + +inline goffset +MadDecoder::ThisFrameOffset() const +{ + goffset offset = input_stream->GetOffset(); + + if (stream.this_frame != nullptr) + offset -= stream.bufend - stream.this_frame; + else + offset -= stream.bufend - stream.buffer; + + return offset; +} + +inline goffset +MadDecoder::RestIncludingThisFrame() const +{ + return input_stream->GetSize() - ThisFrameOffset(); +} + +inline void +MadDecoder::FileSizeToSongLength() +{ + goffset rest = RestIncludingThisFrame(); + + if (rest > 0) { + float frame_duration = mp3_frame_duration(&frame); + + total_time = (rest * 8.0) / frame.header.bitrate; + max_frames = total_time / frame_duration + FRAMES_CUSHION; + } else { + max_frames = FRAMES_CUSHION; + total_time = 0; + } +} + +inline bool +MadDecoder::DecodeFirstFrame(Tag **tag) +{ + struct xing xing; + struct lame lame; + struct mad_bitptr ptr; + int bitlen; + enum mp3_action ret; + + /* stfu gcc */ + memset(&xing, 0, sizeof(struct xing)); + xing.flags = 0; + + while (true) { + do { + ret = DecodeNextFrameHeader(tag); + } while (ret == DECODE_CONT); + if (ret == DECODE_BREAK) + return false; + if (ret == DECODE_SKIP) continue; + + do { + ret = DecodeNextFrame(); + } while (ret == DECODE_CONT); + if (ret == DECODE_BREAK) + return false; + if (ret == DECODE_OK) break; + } + + ptr = stream.anc_ptr; + bitlen = stream.anc_bitlen; + + FileSizeToSongLength(); + + /* + * if an xing tag exists, use that! + */ + if (parse_xing(&xing, &ptr, &bitlen)) { + found_xing = true; + mute_frame = MUTEFRAME_SKIP; + + if ((xing.flags & XING_FRAMES) && xing.frames) { + mad_timer_t duration = frame.header.duration; + mad_timer_multiply(&duration, xing.frames); + total_time = ((float)mad_timer_count(duration, MAD_UNITS_MILLISECONDS)) / 1000; + max_frames = xing.frames; + } + + if (parse_lame(&lame, &ptr, &bitlen)) { + if (gapless_playback && input_stream->IsSeekable()) { + drop_start_samples = lame.encoder_delay + + DECODERDELAY; + drop_end_samples = lame.encoder_padding; + } + + /* Album gain isn't currently used. See comment in + * parse_lame() for details. -- jat */ + if (decoder != nullptr && !found_replay_gain && + lame.track_gain) { + struct replay_gain_info rgi; + replay_gain_info_init(&rgi); + rgi.tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain; + rgi.tuples[REPLAY_GAIN_TRACK].peak = lame.peak; + decoder_replay_gain(decoder, &rgi); + } + } + } + + if (!max_frames) + return false; + + if (max_frames > 8 * 1024 * 1024) { + g_warning("mp3 file header indicates too many frames: %lu\n", + max_frames); + return false; + } + + frame_offsets = new long[max_frames]; + times = new mad_timer_t[max_frames]; + + return true; +} + +MadDecoder::~MadDecoder() +{ + mad_synth_finish(&synth); + mad_frame_finish(&frame); + mad_stream_finish(&stream); + + delete[] frame_offsets; + delete[] times; +} + +/* this is primarily used for getting total time for tags */ +static int +mad_decoder_total_file_time(struct input_stream *is) +{ + MadDecoder data(nullptr, is); + return data.DecodeFirstFrame(nullptr) + ? data.total_time + 0.5 + : -1; +} + +long +MadDecoder::TimeToFrame(double t) const +{ + unsigned long i; + + for (i = 0; i < highest_frame; ++i) { + double frame_time = + mad_timer_count(times[i], + MAD_UNITS_MILLISECONDS) / 1000.; + if (frame_time >= t) + break; + } + + return i; +} + +void +MadDecoder::UpdateTimerNextFrame() +{ + if (current_frame >= highest_frame) { + /* record this frame's properties in frame_offsets + (for seeking) and times */ + bit_rate = frame.header.bitrate; + + if (current_frame >= max_frames) + /* cap current_frame */ + current_frame = max_frames - 1; + else + highest_frame++; + + frame_offsets[current_frame] = ThisFrameOffset(); + + mad_timer_add(&timer, frame.header.duration); + times[current_frame] = timer; + } else + /* get the new timer value from "times" */ + timer = times[current_frame]; + + current_frame++; + elapsed_time = mad_timer_count(timer, MAD_UNITS_MILLISECONDS) / 1000.0; +} + +DecoderCommand +MadDecoder::SendPCM(unsigned i, unsigned pcm_length) +{ + unsigned max_samples; + + max_samples = sizeof(output_buffer) / + sizeof(output_buffer[0]) / + MAD_NCHANNELS(&frame.header); + + while (i < pcm_length) { + unsigned int num_samples = pcm_length - i; + if (num_samples > max_samples) + num_samples = max_samples; + + i += num_samples; + + mad_fixed_to_24_buffer(output_buffer, &synth, + i - num_samples, i, + MAD_NCHANNELS(&frame.header)); + num_samples *= MAD_NCHANNELS(&frame.header); + + auto cmd = decoder_data(decoder, input_stream, output_buffer, + sizeof(output_buffer[0]) * num_samples, + bit_rate / 1000); + if (cmd != DecoderCommand::NONE) + return cmd; + } + + return DecoderCommand::NONE; +} + +inline DecoderCommand +MadDecoder::SyncAndSend() +{ + mad_synth_frame(&synth, &frame); + + if (!found_first_frame) { + unsigned int samples_per_frame = synth.pcm.length; + drop_start_frames = drop_start_samples / samples_per_frame; + drop_end_frames = drop_end_samples / samples_per_frame; + drop_start_samples = drop_start_samples % samples_per_frame; + drop_end_samples = drop_end_samples % samples_per_frame; + found_first_frame = true; + } + + if (drop_start_frames > 0) { + drop_start_frames--; + return DecoderCommand::NONE; + } else if ((drop_end_frames > 0) && + (current_frame == (max_frames + 1 - drop_end_frames))) { + /* stop decoding, effectively dropping all remaining + frames */ + return DecoderCommand::STOP; + } + + unsigned i = 0; + if (!decoded_first_frame) { + i = drop_start_samples; + decoded_first_frame = true; + } + + unsigned pcm_length = synth.pcm.length; + if (drop_end_samples && + (current_frame == max_frames - drop_end_frames)) { + if (drop_end_samples >= pcm_length) + pcm_length = 0; + else + pcm_length -= drop_end_samples; + } + + auto cmd = SendPCM(i, pcm_length); + if (cmd != DecoderCommand::NONE) + return cmd; + + if (drop_end_samples && + (current_frame == max_frames - drop_end_frames)) + /* stop decoding, effectively dropping + * all remaining samples */ + return DecoderCommand::STOP; + + return DecoderCommand::NONE; +} + +inline bool +MadDecoder::Read() +{ + enum mp3_action ret; + + UpdateTimerNextFrame(); + + switch (mute_frame) { + DecoderCommand cmd; + + case MUTEFRAME_SKIP: + mute_frame = MUTEFRAME_NONE; + break; + case MUTEFRAME_SEEK: + if (elapsed_time >= seek_where) + mute_frame = MUTEFRAME_NONE; + break; + case MUTEFRAME_NONE: + cmd = SyncAndSend(); + if (cmd == DecoderCommand::SEEK) { + unsigned long j; + + assert(input_stream->IsSeekable()); + + j = TimeToFrame(decoder_seek_where(decoder)); + if (j < highest_frame) { + if (Seek(frame_offsets[j])) { + current_frame = j; + decoder_command_finished(decoder); + } else + decoder_seek_error(decoder); + } else { + seek_where = decoder_seek_where(decoder); + mute_frame = MUTEFRAME_SEEK; + decoder_command_finished(decoder); + } + } else if (cmd != DecoderCommand::NONE) + return false; + } + + while (true) { + bool skip = false; + + do { + Tag *tag = nullptr; + + ret = DecodeNextFrameHeader(&tag); + + if (tag != nullptr) { + decoder_tag(decoder, input_stream, + std::move(*tag)); + delete tag; + } + } while (ret == DECODE_CONT); + if (ret == DECODE_BREAK) + return false; + else if (ret == DECODE_SKIP) + skip = true; + + if (mute_frame == MUTEFRAME_NONE) { + do { + ret = DecodeNextFrame(); + } while (ret == DECODE_CONT); + if (ret == DECODE_BREAK) + return false; + } + + if (!skip && ret == DECODE_OK) + break; + } + + return ret != DECODE_BREAK; +} + +static void +mp3_decode(struct decoder *decoder, struct input_stream *input_stream) +{ + MadDecoder data(decoder, input_stream); + + Tag *tag = nullptr; + if (!data.DecodeFirstFrame(&tag)) { + delete tag; + + if (decoder_get_command(decoder) == DecoderCommand::NONE) + g_warning + ("Input does not appear to be a mp3 bit stream.\n"); + return; + } + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, + data.frame.header.samplerate, + SampleFormat::S24_P32, + MAD_NCHANNELS(&data.frame.header), + error)) { + g_warning("%s", error.GetMessage()); + delete tag; + return; + } + + decoder_initialized(decoder, audio_format, + input_stream->IsSeekable(), + data.total_time); + + if (tag != nullptr) { + decoder_tag(decoder, input_stream, std::move(*tag)); + delete tag; + } + + while (data.Read()) {} +} + +static bool +mad_decoder_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + int total_time; + + total_time = mad_decoder_total_file_time(is); + if (total_time < 0) + return false; + + tag_handler_invoke_duration(handler, handler_ctx, total_time); + return true; +} + +static const char *const mp3_suffixes[] = { "mp3", "mp2", nullptr }; +static const char *const mp3_mime_types[] = { "audio/mpeg", nullptr }; + +const struct decoder_plugin mad_decoder_plugin = { + "mad", + mp3_plugin_init, + nullptr, + mp3_decode, + nullptr, + nullptr, + mad_decoder_scan_stream, + nullptr, + mp3_suffixes, + mp3_mime_types, +}; diff --git a/src/decoder/MadDecoderPlugin.hxx b/src/decoder/MadDecoderPlugin.hxx new file mode 100644 index 000000000..c7a77304c --- /dev/null +++ b/src/decoder/MadDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_MAD_HXX +#define MPD_DECODER_MAD_HXX + +extern const struct decoder_plugin mad_decoder_plugin; + +#endif diff --git a/src/decoder/MikmodDecoderPlugin.cxx b/src/decoder/MikmodDecoderPlugin.cxx new file mode 100644 index 000000000..78a26891a --- /dev/null +++ b/src/decoder/MikmodDecoderPlugin.cxx @@ -0,0 +1,243 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "MikmodDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "tag/TagHandler.hxx" +#include "system/FatalError.hxx" + +#include <glib.h> +#include <mikmod.h> +#include <assert.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "mikmod" + +/* this is largely copied from alsaplayer */ + +static constexpr size_t MIKMOD_FRAME_SIZE = 4096; + +static BOOL +mikmod_mpd_init(void) +{ + return VC_Init(); +} + +static void +mikmod_mpd_exit(void) +{ + VC_Exit(); +} + +static void +mikmod_mpd_update(void) +{ +} + +static BOOL +mikmod_mpd_is_present(void) +{ + return true; +} + +static char drv_name[] = PACKAGE_NAME; +static char drv_version[] = VERSION; + +#if (LIBMIKMOD_VERSION > 0x030106) +static char drv_alias[] = PACKAGE; +#endif + +static MDRIVER drv_mpd = { + nullptr, + drv_name, + drv_version, + 0, + 255, +#if (LIBMIKMOD_VERSION > 0x030106) + drv_alias, +#if (LIBMIKMOD_VERSION >= 0x030200) + nullptr, /* CmdLineHelp */ +#endif + nullptr, /* CommandLine */ +#endif + mikmod_mpd_is_present, + VC_SampleLoad, + VC_SampleUnload, + VC_SampleSpace, + VC_SampleLength, + mikmod_mpd_init, + mikmod_mpd_exit, + nullptr, + VC_SetNumVoices, + VC_PlayStart, + VC_PlayStop, + mikmod_mpd_update, + nullptr, + VC_VoiceSetVolume, + VC_VoiceGetVolume, + VC_VoiceSetFrequency, + VC_VoiceGetFrequency, + VC_VoiceSetPanning, + VC_VoiceGetPanning, + VC_VoicePlay, + VC_VoiceStop, + VC_VoiceStopped, + VC_VoiceGetPosition, + VC_VoiceRealVolume +}; + +static unsigned mikmod_sample_rate; + +static bool +mikmod_decoder_init(const config_param ¶m) +{ + static char params[] = ""; + + mikmod_sample_rate = param.GetBlockValue("sample_rate", 44100u); + if (!audio_valid_sample_rate(mikmod_sample_rate)) + FormatFatalError("Invalid sample rate in line %d: %u", + param.line, mikmod_sample_rate); + + md_device = 0; + md_reverb = 0; + + MikMod_RegisterDriver(&drv_mpd); + MikMod_RegisterAllLoaders(); + + md_pansep = 64; + md_mixfreq = mikmod_sample_rate; + md_mode = (DMODE_SOFT_MUSIC | DMODE_INTERP | DMODE_STEREO | + DMODE_16BITS); + + if (MikMod_Init(params)) { + g_warning("Could not init MikMod: %s\n", + MikMod_strerror(MikMod_errno)); + return false; + } + + return true; +} + +static void +mikmod_decoder_finish(void) +{ + MikMod_Exit(); +} + +static void +mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs) +{ + char *path2; + MODULE *handle; + int ret; + SBYTE buffer[MIKMOD_FRAME_SIZE]; + + path2 = g_strdup(path_fs); + handle = Player_Load(path2, 128, 0); + g_free(path2); + + if (handle == nullptr) { + g_warning("failed to open mod: %s", path_fs); + return; + } + + /* Prevent module from looping forever */ + handle->loop = 0; + + const AudioFormat audio_format(mikmod_sample_rate, SampleFormat::S16, 2); + assert(audio_format.IsValid()); + + decoder_initialized(decoder, audio_format, false, 0); + + Player_Start(handle); + + DecoderCommand cmd = DecoderCommand::NONE; + while (cmd == DecoderCommand::NONE && Player_Active()) { + ret = VC_WriteBytes(buffer, sizeof(buffer)); + cmd = decoder_data(decoder, nullptr, buffer, ret, 0); + } + + Player_Stop(); + Player_Free(handle); +} + +static bool +mikmod_decoder_scan_file(const char *path_fs, + const struct tag_handler *handler, void *handler_ctx) +{ + char *path2 = g_strdup(path_fs); + MODULE *handle = Player_Load(path2, 128, 0); + + if (handle == nullptr) { + g_free(path2); + g_debug("Failed to open file: %s", path_fs); + return false; + + } + + Player_Free(handle); + + char *title = Player_LoadTitle(path2); + g_free(path2); + + if (title != nullptr) { + tag_handler_invoke_tag(handler, handler_ctx, + TAG_TITLE, title); +#if (LIBMIKMOD_VERSION >= 0x030200) + MikMod_free(title); +#else + free(title); +#endif + } + + return true; +} + +static const char *const mikmod_decoder_suffixes[] = { + "amf", + "dsm", + "far", + "gdm", + "imf", + "it", + "med", + "mod", + "mtm", + "s3m", + "stm", + "stx", + "ult", + "uni", + "xm", + nullptr +}; + +const struct decoder_plugin mikmod_decoder_plugin = { + "mikmod", + mikmod_decoder_init, + mikmod_decoder_finish, + nullptr, + mikmod_decoder_file_decode, + mikmod_decoder_scan_file, + nullptr, + nullptr, + mikmod_decoder_suffixes, + nullptr, +}; diff --git a/src/decoder/MikmodDecoderPlugin.hxx b/src/decoder/MikmodDecoderPlugin.hxx new file mode 100644 index 000000000..dd3b1389e --- /dev/null +++ b/src/decoder/MikmodDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_MIKMOD_HXX +#define MPD_DECODER_MIKMOD_HXX + +extern const struct decoder_plugin mikmod_decoder_plugin; + +#endif diff --git a/src/decoder/ModplugDecoderPlugin.cxx b/src/decoder/ModplugDecoderPlugin.cxx new file mode 100644 index 000000000..9cbf44b15 --- /dev/null +++ b/src/decoder/ModplugDecoderPlugin.cxx @@ -0,0 +1,203 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "ModplugDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "tag/TagHandler.hxx" + +#include <libmodplug/modplug.h> + +#include <glib.h> + +#include <assert.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "modplug" + +static constexpr size_t MODPLUG_FRAME_SIZE = 4096; +static constexpr size_t MODPLUG_PREALLOC_BLOCK = 256 * 1024; +static constexpr size_t MODPLUG_READ_BLOCK = 128 * 1024; +static constexpr goffset MODPLUG_FILE_LIMIT = 100 * 1024 * 1024; + +static GByteArray * +mod_loadfile(struct decoder *decoder, struct input_stream *is) +{ + const goffset size = is->GetSize(); + + if (size == 0) { + g_warning("file is empty"); + return nullptr; + } + + if (size > MODPLUG_FILE_LIMIT) { + g_warning("file too large"); + return nullptr; + } + + //known/unknown size, preallocate array, lets read in chunks + GByteArray *bdatas; + if (size > 0) { + bdatas = g_byte_array_sized_new(size); + } else { + bdatas = g_byte_array_sized_new(MODPLUG_PREALLOC_BLOCK); + } + + unsigned char *data = (unsigned char *)g_malloc(MODPLUG_READ_BLOCK); + + while (true) { + size_t ret = decoder_read(decoder, is, data, + MODPLUG_READ_BLOCK); + if (ret == 0) { + if (is->LockIsEOF()) + /* end of file */ + break; + + /* I/O error - skip this song */ + g_free(data); + g_byte_array_free(bdatas, true); + return nullptr; + } + + if (goffset(bdatas->len + ret) > MODPLUG_FILE_LIMIT) { + g_warning("stream too large\n"); + g_free(data); + g_byte_array_free(bdatas, TRUE); + return nullptr; + } + + g_byte_array_append(bdatas, data, ret); + } + + g_free(data); + + return bdatas; +} + +static void +mod_decode(struct decoder *decoder, struct input_stream *is) +{ + ModPlugFile *f; + ModPlug_Settings settings; + GByteArray *bdatas; + int ret; + char audio_buffer[MODPLUG_FRAME_SIZE]; + + bdatas = mod_loadfile(decoder, is); + + if (!bdatas) { + g_warning("could not load stream\n"); + return; + } + + ModPlug_GetSettings(&settings); + /* alter setting */ + settings.mResamplingMode = MODPLUG_RESAMPLE_FIR; /* RESAMP */ + settings.mChannels = 2; + settings.mBits = 16; + settings.mFrequency = 44100; + /* insert more setting changes here */ + ModPlug_SetSettings(&settings); + + f = ModPlug_Load(bdatas->data, bdatas->len); + g_byte_array_free(bdatas, TRUE); + if (!f) { + g_warning("could not decode stream\n"); + return; + } + + static constexpr AudioFormat audio_format(44100, SampleFormat::S16, 2); + assert(audio_format.IsValid()); + + decoder_initialized(decoder, audio_format, + is->IsSeekable(), + ModPlug_GetLength(f) / 1000.0); + + DecoderCommand cmd; + do { + ret = ModPlug_Read(f, audio_buffer, MODPLUG_FRAME_SIZE); + if (ret <= 0) + break; + + cmd = decoder_data(decoder, nullptr, + audio_buffer, ret, + 0); + + if (cmd == DecoderCommand::SEEK) { + float where = decoder_seek_where(decoder); + + ModPlug_Seek(f, (int)(where * 1000.0)); + + decoder_command_finished(decoder); + } + + } while (cmd != DecoderCommand::STOP); + + ModPlug_Unload(f); +} + +static bool +modplug_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + ModPlugFile *f; + GByteArray *bdatas; + + bdatas = mod_loadfile(nullptr, is); + if (!bdatas) + return false; + + f = ModPlug_Load(bdatas->data, bdatas->len); + g_byte_array_free(bdatas, TRUE); + if (f == nullptr) + return false; + + tag_handler_invoke_duration(handler, handler_ctx, + ModPlug_GetLength(f) / 1000); + + const char *title = ModPlug_GetName(f); + if (title != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_TITLE, title); + + ModPlug_Unload(f); + + return true; +} + +static const char *const mod_suffixes[] = { + "669", "amf", "ams", "dbm", "dfm", "dsm", "far", "it", + "med", "mdl", "mod", "mtm", "mt2", "okt", "s3m", "stm", + "ult", "umx", "xm", + nullptr +}; + +const struct decoder_plugin modplug_decoder_plugin = { + "modplug", + nullptr, + nullptr, + mod_decode, + nullptr, + nullptr, + modplug_scan_stream, + nullptr, + mod_suffixes, + nullptr, +}; diff --git a/src/decoder/ModplugDecoderPlugin.hxx b/src/decoder/ModplugDecoderPlugin.hxx new file mode 100644 index 000000000..fefb02b05 --- /dev/null +++ b/src/decoder/ModplugDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_MODPLUG_HXX +#define MPD_DECODER_MODPLUG_HXX + +extern const struct decoder_plugin modplug_decoder_plugin; + +#endif diff --git a/src/decoder/MpcdecDecoderPlugin.cxx b/src/decoder/MpcdecDecoderPlugin.cxx new file mode 100644 index 000000000..252fe92e6 --- /dev/null +++ b/src/decoder/MpcdecDecoderPlugin.cxx @@ -0,0 +1,278 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "MpcdecDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "util/Error.hxx" + +#include <mpc/mpcdec.h> + +#include <glib.h> +#include <assert.h> +#include <unistd.h> +#include <math.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "mpcdec" + +struct mpc_decoder_data { + struct input_stream *is; + struct decoder *decoder; +}; + +static mpc_int32_t +mpc_read_cb(mpc_reader *reader, void *ptr, mpc_int32_t size) +{ + struct mpc_decoder_data *data = + (struct mpc_decoder_data *)reader->data; + + return decoder_read(data->decoder, data->is, ptr, size); +} + +static mpc_bool_t +mpc_seek_cb(mpc_reader *reader, mpc_int32_t offset) +{ + struct mpc_decoder_data *data = + (struct mpc_decoder_data *)reader->data; + + return data->is->LockSeek(offset, SEEK_SET, IgnoreError()); +} + +static mpc_int32_t +mpc_tell_cb(mpc_reader *reader) +{ + struct mpc_decoder_data *data = + (struct mpc_decoder_data *)reader->data; + + return (long)data->is->GetOffset(); +} + +static mpc_bool_t +mpc_canseek_cb(mpc_reader *reader) +{ + struct mpc_decoder_data *data = + (struct mpc_decoder_data *)reader->data; + + return data->is->IsSeekable(); +} + +static mpc_int32_t +mpc_getsize_cb(mpc_reader *reader) +{ + struct mpc_decoder_data *data = + (struct mpc_decoder_data *)reader->data; + + return data->is->GetSize(); +} + +/* this _looks_ performance-critical, don't de-inline -- eric */ +static inline int32_t +mpc_to_mpd_sample(MPC_SAMPLE_FORMAT sample) +{ + /* only doing 16-bit audio for now */ + int32_t val; + + enum { + bits = 24, + }; + + const int clip_min = -1 << (bits - 1); + const int clip_max = (1 << (bits - 1)) - 1; + +#ifdef MPC_FIXED_POINT + const int shift = bits - MPC_FIXED_POINT_SCALE_SHIFT; + + if (shift < 0) + val = sample >> -shift; + else + val = sample << shift; +#else + const int float_scale = 1 << (bits - 1); + + val = sample * float_scale; +#endif + + if (val < clip_min) + val = clip_min; + else if (val > clip_max) + val = clip_max; + + return val; +} + +static void +mpc_to_mpd_buffer(int32_t *dest, const MPC_SAMPLE_FORMAT *src, + unsigned num_samples) +{ + while (num_samples-- > 0) + *dest++ = mpc_to_mpd_sample(*src++); +} + +static void +mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) +{ + MPC_SAMPLE_FORMAT sample_buffer[MPC_DECODER_BUFFER_LENGTH]; + + struct mpc_decoder_data data; + data.is = is; + data.decoder = mpd_decoder; + + mpc_reader reader; + reader.read = mpc_read_cb; + reader.seek = mpc_seek_cb; + reader.tell = mpc_tell_cb; + reader.get_size = mpc_getsize_cb; + reader.canseek = mpc_canseek_cb; + reader.data = &data; + + mpc_demux *demux = mpc_demux_init(&reader); + if (demux == nullptr) { + if (decoder_get_command(mpd_decoder) != DecoderCommand::STOP) + g_warning("Not a valid musepack stream"); + return; + } + + mpc_streaminfo info; + mpc_demux_get_info(demux, &info); + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, info.sample_freq, + SampleFormat::S24_P32, + info.channels, error)) { + g_warning("%s", error.GetMessage()); + mpc_demux_exit(demux); + return; + } + + struct replay_gain_info replay_gain_info; + replay_gain_info_init(&replay_gain_info); + replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = MPC_OLD_GAIN_REF - (info.gain_album / 256.); + replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = pow(10, info.peak_album / 256. / 20) / 32767; + replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = MPC_OLD_GAIN_REF - (info.gain_title / 256.); + replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = pow(10, info.peak_title / 256. / 20) / 32767; + + decoder_replay_gain(mpd_decoder, &replay_gain_info); + + decoder_initialized(mpd_decoder, audio_format, + is->IsSeekable(), + mpc_streaminfo_get_length(&info)); + + DecoderCommand cmd = DecoderCommand::NONE; + do { + if (cmd == DecoderCommand::SEEK) { + mpc_int64_t where = decoder_seek_where(mpd_decoder) * + audio_format.sample_rate; + bool success; + + success = mpc_demux_seek_sample(demux, where) + == MPC_STATUS_OK; + if (success) + decoder_command_finished(mpd_decoder); + else + decoder_seek_error(mpd_decoder); + } + + mpc_uint32_t vbr_update_bits = 0; + + mpc_frame_info frame; + frame.buffer = (MPC_SAMPLE_FORMAT *)sample_buffer; + mpc_status status = mpc_demux_decode(demux, &frame); + if (status != MPC_STATUS_OK) { + g_warning("Failed to decode sample"); + break; + } + + if (frame.bits == -1) + break; + + mpc_uint32_t ret = frame.samples; + ret *= info.channels; + + int32_t chunk[G_N_ELEMENTS(sample_buffer)]; + mpc_to_mpd_buffer(chunk, sample_buffer, ret); + + long bit_rate = vbr_update_bits * audio_format.sample_rate + / 1152 / 1000; + + cmd = decoder_data(mpd_decoder, is, + chunk, ret * sizeof(chunk[0]), + bit_rate); + } while (cmd != DecoderCommand::STOP); + + mpc_demux_exit(demux); +} + +static float +mpcdec_get_file_duration(struct input_stream *is) +{ + struct mpc_decoder_data data; + data.is = is; + data.decoder = nullptr; + + mpc_reader reader; + reader.read = mpc_read_cb; + reader.seek = mpc_seek_cb; + reader.tell = mpc_tell_cb; + reader.get_size = mpc_getsize_cb; + reader.canseek = mpc_canseek_cb; + reader.data = &data; + + mpc_demux *demux = mpc_demux_init(&reader); + if (demux == nullptr) + return -1; + + mpc_streaminfo info; + mpc_demux_get_info(demux, &info); + mpc_demux_exit(demux); + + return mpc_streaminfo_get_length(&info); +} + +static bool +mpcdec_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + float total_time = mpcdec_get_file_duration(is); + + if (total_time < 0) + return false; + + tag_handler_invoke_duration(handler, handler_ctx, total_time); + return true; +} + +static const char *const mpcdec_suffixes[] = { "mpc", nullptr }; + +const struct decoder_plugin mpcdec_decoder_plugin = { + "mpcdec", + nullptr, + nullptr, + mpcdec_decode, + nullptr, + nullptr, + mpcdec_scan_stream, + nullptr, + mpcdec_suffixes, + nullptr, +}; diff --git a/src/decoder/MpcdecDecoderPlugin.hxx b/src/decoder/MpcdecDecoderPlugin.hxx new file mode 100644 index 000000000..7e9b51cdb --- /dev/null +++ b/src/decoder/MpcdecDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_MPCDEC_HXX +#define MPD_DECODER_MPCDEC_HXX + +extern const struct decoder_plugin mpcdec_decoder_plugin; + +#endif diff --git a/src/decoder/Mpg123DecoderPlugin.cxx b/src/decoder/Mpg123DecoderPlugin.cxx new file mode 100644 index 000000000..3100a0f1c --- /dev/null +++ b/src/decoder/Mpg123DecoderPlugin.cxx @@ -0,0 +1,250 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" /* must be first for large file support */ +#include "Mpg123DecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "util/Error.hxx" + +#include <glib.h> + +#include <mpg123.h> +#include <stdio.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "mpg123" + +static bool +mpd_mpg123_init(gcc_unused const config_param ¶m) +{ + mpg123_init(); + + return true; +} + +static void +mpd_mpg123_finish(void) +{ + mpg123_exit(); +} + +/** + * Opens a file with an existing #mpg123_handle. + * + * @param handle a handle which was created before; on error, this + * function will not free it + * @param audio_format this parameter is filled after successful + * return + * @return true on success + */ +static bool +mpd_mpg123_open(mpg123_handle *handle, const char *path_fs, + AudioFormat &audio_format) +{ + char *path_dup; + int error; + int channels, encoding; + long rate; + + /* mpg123_open() wants a writable string :-( */ + path_dup = g_strdup(path_fs); + + error = mpg123_open(handle, path_dup); + g_free(path_dup); + if (error != MPG123_OK) { + g_warning("libmpg123 failed to open %s: %s", + path_fs, mpg123_plain_strerror(error)); + return false; + } + + /* obtain the audio format */ + + error = mpg123_getformat(handle, &rate, &channels, &encoding); + if (error != MPG123_OK) { + g_warning("mpg123_getformat() failed: %s", + mpg123_plain_strerror(error)); + return false; + } + + if (encoding != MPG123_ENC_SIGNED_16) { + /* other formats not yet implemented */ + g_warning("expected MPG123_ENC_SIGNED_16, got %d", encoding); + return false; + } + + Error error2; + if (!audio_format_init_checked(audio_format, rate, SampleFormat::S16, + channels, error2)) { + g_warning("%s", error2.GetMessage()); + return false; + } + + return true; +} + +static void +mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs) +{ + mpg123_handle *handle; + int error; + off_t num_samples; + struct mpg123_frameinfo info; + + /* open the file */ + + handle = mpg123_new(nullptr, &error); + if (handle == nullptr) { + g_warning("mpg123_new() failed: %s", + mpg123_plain_strerror(error)); + return; + } + + AudioFormat audio_format; + if (!mpd_mpg123_open(handle, path_fs, audio_format)) { + mpg123_delete(handle); + return; + } + + num_samples = mpg123_length(handle); + + /* tell MPD core we're ready */ + + decoder_initialized(decoder, audio_format, true, + (float)num_samples / + (float)audio_format.sample_rate); + + if (mpg123_info(handle, &info) != MPG123_OK) { + info.vbr = MPG123_CBR; + info.bitrate = 0; + } + + switch (info.vbr) { + case MPG123_ABR: + info.bitrate = info.abr_rate; + break; + case MPG123_CBR: + break; + default: + info.bitrate = 0; + } + + /* the decoder main loop */ + + DecoderCommand cmd; + do { + unsigned char buffer[8192]; + size_t nbytes; + + /* decode */ + + error = mpg123_read(handle, buffer, sizeof(buffer), &nbytes); + if (error != MPG123_OK) { + if (error != MPG123_DONE) + g_warning("mpg123_read() failed: %s", + mpg123_plain_strerror(error)); + break; + } + + /* update bitrate for ABR/VBR */ + if (info.vbr != MPG123_CBR) { + /* FIXME: maybe skip, as too expensive? */ + /* FIXME: maybe, (info.vbr == MPG123_VBR) ? */ + if (mpg123_info (handle, &info) != MPG123_OK) + info.bitrate = 0; + } + + /* send to MPD */ + + cmd = decoder_data(decoder, nullptr, buffer, nbytes, info.bitrate); + + if (cmd == DecoderCommand::SEEK) { + off_t c = decoder_seek_where(decoder)*audio_format.sample_rate; + c = mpg123_seek(handle, c, SEEK_SET); + if (c < 0) + decoder_seek_error(decoder); + else { + decoder_command_finished(decoder); + decoder_timestamp(decoder, c/(double)audio_format.sample_rate); + } + + cmd = DecoderCommand::NONE; + } + } while (cmd == DecoderCommand::NONE); + + /* cleanup */ + + mpg123_delete(handle); +} + +static bool +mpd_mpg123_scan_file(const char *path_fs, + const struct tag_handler *handler, void *handler_ctx) +{ + mpg123_handle *handle; + int error; + off_t num_samples; + + handle = mpg123_new(nullptr, &error); + if (handle == nullptr) { + g_warning("mpg123_new() failed: %s", + mpg123_plain_strerror(error)); + return false; + } + + AudioFormat audio_format; + if (!mpd_mpg123_open(handle, path_fs, audio_format)) { + mpg123_delete(handle); + return false; + } + + num_samples = mpg123_length(handle); + if (num_samples <= 0) { + mpg123_delete(handle); + return false; + } + + /* ID3 tag support not yet implemented */ + + mpg123_delete(handle); + + tag_handler_invoke_duration(handler, handler_ctx, + num_samples / audio_format.sample_rate); + return true; +} + +static const char *const mpg123_suffixes[] = { + "mp3", + nullptr +}; + +const struct decoder_plugin mpg123_decoder_plugin = { + "mpg123", + mpd_mpg123_init, + mpd_mpg123_finish, + /* streaming not yet implemented */ + nullptr, + mpd_mpg123_file_decode, + mpd_mpg123_scan_file, + nullptr, + nullptr, + mpg123_suffixes, + nullptr, +}; diff --git a/src/decoder/Mpg123DecoderPlugin.hxx b/src/decoder/Mpg123DecoderPlugin.hxx new file mode 100644 index 000000000..273b03eaf --- /dev/null +++ b/src/decoder/Mpg123DecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_MPG123_HXX +#define MPD_DECODER_MPG123_HXX + +extern const struct decoder_plugin mpg123_decoder_plugin; + +#endif diff --git a/src/decoder/OggCodec.cxx b/src/decoder/OggCodec.cxx new file mode 100644 index 000000000..d7e5b7642 --- /dev/null +++ b/src/decoder/OggCodec.cxx @@ -0,0 +1,50 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* + * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC) + */ + +#include "config.h" +#include "OggCodec.hxx" + +#include <string.h> + +enum ogg_codec +ogg_codec_detect(struct decoder *decoder, struct input_stream *is) +{ + /* oggflac detection based on code in ogg123 and this post + * http://lists.xiph.org/pipermail/flac/2004-December/000393.html + * ogg123 trunk still doesn't have this patch as of June 2005 */ + unsigned char buf[41]; + size_t r = decoder_read(decoder, is, buf, sizeof(buf)); + if (r < sizeof(buf) || memcmp(buf, "OggS", 4) != 0) + return OGG_CODEC_UNKNOWN; + + if ((memcmp(buf + 29, "FLAC", 4) == 0 && + memcmp(buf + 37, "fLaC", 4) == 0) || + memcmp(buf + 28, "FLAC", 4) == 0 || + memcmp(buf + 28, "fLaC", 4) == 0) + return OGG_CODEC_FLAC; + + if (memcmp(buf + 28, "Opus", 4) == 0) + return OGG_CODEC_OPUS; + + return OGG_CODEC_VORBIS; +} diff --git a/src/decoder/OggCodec.hxx b/src/decoder/OggCodec.hxx new file mode 100644 index 000000000..eb709286b --- /dev/null +++ b/src/decoder/OggCodec.hxx @@ -0,0 +1,39 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/* + * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC) + */ + +#ifndef MPD_OGG_CODEC_HXX +#define MPD_OGG_CODEC_HXX + +#include "DecoderAPI.hxx" + +enum ogg_codec { + OGG_CODEC_UNKNOWN, + OGG_CODEC_VORBIS, + OGG_CODEC_FLAC, + OGG_CODEC_OPUS, +}; + +enum ogg_codec +ogg_codec_detect(struct decoder *decoder, struct input_stream *is); + +#endif /* _OGG_COMMON_H */ diff --git a/src/decoder/OggFind.cxx b/src/decoder/OggFind.cxx new file mode 100644 index 000000000..9df4c6455 --- /dev/null +++ b/src/decoder/OggFind.cxx @@ -0,0 +1,37 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "OggFind.hxx" +#include "OggSyncState.hxx" + +bool +OggFindEOS(OggSyncState &oy, ogg_stream_state &os, ogg_packet &packet) +{ + while (true) { + int r = ogg_stream_packetout(&os, &packet); + if (r == 0) { + if (!oy.ExpectPageIn(os)) + return false; + + continue; + } else if (r > 0 && packet.e_o_s) + return true; + } +} diff --git a/src/decoder/OggFind.hxx b/src/decoder/OggFind.hxx new file mode 100644 index 000000000..7d18d2067 --- /dev/null +++ b/src/decoder/OggFind.hxx @@ -0,0 +1,38 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_OGG_FIND_HXX +#define MPD_OGG_FIND_HXX + +#include "check.h" + +#include <ogg/ogg.h> + +class OggSyncState; + +/** + * Skip all pages/packets until an end-of-stream (EOS) packet for the + * specified stream is found. + * + * @return true if the EOS packet was found + */ +bool +OggFindEOS(OggSyncState &oy, ogg_stream_state &os, ogg_packet &packet); + +#endif diff --git a/src/decoder/OggSyncState.hxx b/src/decoder/OggSyncState.hxx new file mode 100644 index 000000000..eaeb9bd8c --- /dev/null +++ b/src/decoder/OggSyncState.hxx @@ -0,0 +1,78 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_OGG_SYNC_STATE_HXX +#define MPD_OGG_SYNC_STATE_HXX + +#include "check.h" +#include "OggUtil.hxx" + +#include <ogg/ogg.h> + +#include <stddef.h> + +/** + * Wrapper for an ogg_sync_state. + */ +class OggSyncState { + ogg_sync_state oy; + + input_stream &is; + struct decoder *const decoder; + +public: + OggSyncState(input_stream &_is, struct decoder *const _decoder=nullptr) + :is(_is), decoder(_decoder) { + ogg_sync_init(&oy); + } + + ~OggSyncState() { + ogg_sync_clear(&oy); + } + + void Reset() { + ogg_sync_reset(&oy); + } + + bool Feed(size_t size) { + return OggFeed(oy, decoder, &is, size); + } + + bool ExpectPage(ogg_page &page) { + return OggExpectPage(oy, page, decoder, &is); + } + + bool ExpectFirstPage(ogg_stream_state &os) { + return OggExpectFirstPage(oy, os, decoder, &is); + } + + bool ExpectPageIn(ogg_stream_state &os) { + return OggExpectPageIn(oy, os, decoder, &is); + } + + bool ExpectPageSeek(ogg_page &page) { + return OggExpectPageSeek(oy, page, decoder, &is); + } + + bool ExpectPageSeekIn(ogg_stream_state &os) { + return OggExpectPageSeekIn(oy, os, decoder, &is); + } +}; + +#endif diff --git a/src/decoder/OggUtil.cxx b/src/decoder/OggUtil.cxx new file mode 100644 index 000000000..0e2f48f51 --- /dev/null +++ b/src/decoder/OggUtil.cxx @@ -0,0 +1,118 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "OggUtil.hxx" +#include "DecoderAPI.hxx" + +bool +OggFeed(ogg_sync_state &oy, struct decoder *decoder, + input_stream *input_stream, size_t size) +{ + char *buffer = ogg_sync_buffer(&oy, size); + if (buffer == nullptr) + return false; + + size_t nbytes = decoder_read(decoder, input_stream, + buffer, size); + if (nbytes == 0) + return false; + + ogg_sync_wrote(&oy, nbytes); + return true; +} + +bool +OggExpectPage(ogg_sync_state &oy, ogg_page &page, + decoder *decoder, input_stream *input_stream) +{ + while (true) { + int r = ogg_sync_pageout(&oy, &page); + if (r != 0) + return r > 0; + + if (!OggFeed(oy, decoder, input_stream, 1024)) + return false; + } +} + +bool +OggExpectFirstPage(ogg_sync_state &oy, ogg_stream_state &os, + decoder *decoder, input_stream *is) +{ + ogg_page page; + if (!OggExpectPage(oy, page, decoder, is)) + return false; + + ogg_stream_init(&os, ogg_page_serialno(&page)); + ogg_stream_pagein(&os, &page); + return true; +} + +bool +OggExpectPageIn(ogg_sync_state &oy, ogg_stream_state &os, + decoder *decoder, input_stream *is) +{ + ogg_page page; + if (!OggExpectPage(oy, page, decoder, is)) + return false; + + ogg_stream_pagein(&os, &page); + return true; +} + +bool +OggExpectPageSeek(ogg_sync_state &oy, ogg_page &page, + decoder *decoder, input_stream *input_stream) +{ + size_t remaining_skipped = 16384; + + while (true) { + int r = ogg_sync_pageseek(&oy, &page); + if (r > 0) + return true; + + if (r < 0) { + /* skipped -r bytes */ + size_t nbytes = -r; + if (nbytes > remaining_skipped) + /* still no ogg page - we lost our + patience, abort */ + return false; + + remaining_skipped -= nbytes; + continue; + } + + if (!OggFeed(oy, decoder, input_stream, 1024)) + return false; + } +} + +bool +OggExpectPageSeekIn(ogg_sync_state &oy, ogg_stream_state &os, + decoder *decoder, input_stream *is) +{ + ogg_page page; + if (!OggExpectPageSeek(oy, page, decoder, is)) + return false; + + ogg_stream_pagein(&os, &page); + return true; +} diff --git a/src/decoder/OggUtil.hxx b/src/decoder/OggUtil.hxx new file mode 100644 index 000000000..324797815 --- /dev/null +++ b/src/decoder/OggUtil.hxx @@ -0,0 +1,87 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_OGG_UTIL_HXX +#define MPD_OGG_UTIL_HXX + +#include "check.h" + +#include <ogg/ogg.h> + +#include <stddef.h> + +struct input_stream; +struct decoder; + +/** + * Feed data from the #input_stream into the #ogg_sync_state. + * + * @return false on error or end-of-file + */ +bool +OggFeed(ogg_sync_state &oy, struct decoder *decoder, input_stream *is, + size_t size); + +/** + * Feed into the #ogg_sync_state until a page gets available. Garbage + * data at the beginning is considered a fatal error. + * + * @return true if a page is available + */ +bool +OggExpectPage(ogg_sync_state &oy, ogg_page &page, + decoder *decoder, input_stream *input_stream); + +/** + * Combines OggExpectPage(), ogg_stream_init() and + * ogg_stream_pagein(). + * + * @return true if the stream was initialized and the first page was + * delivered to it + */ +bool +OggExpectFirstPage(ogg_sync_state &oy, ogg_stream_state &os, + decoder *decoder, input_stream *is); + +/** + * Combines OggExpectPage() and ogg_stream_pagein(). + * + * @return true if a page was delivered to the stream + */ +bool +OggExpectPageIn(ogg_sync_state &oy, ogg_stream_state &os, + decoder *decoder, input_stream *is); + +/** + * Like OggExpectPage(), but allow skipping garbage (after seeking). + */ +bool +OggExpectPageSeek(ogg_sync_state &oy, ogg_page &page, + decoder *decoder, input_stream *input_stream); + +/** + * Combines OggExpectPageSeek() and ogg_stream_pagein(). + * + * @return true if a page was delivered to the stream + */ +bool +OggExpectPageSeekIn(ogg_sync_state &oy, ogg_stream_state &os, + decoder *decoder, input_stream *is); + +#endif diff --git a/src/decoder/OpusDecoderPlugin.cxx b/src/decoder/OpusDecoderPlugin.cxx new file mode 100644 index 000000000..ea1d6660c --- /dev/null +++ b/src/decoder/OpusDecoderPlugin.cxx @@ -0,0 +1,404 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" /* must be first for large file support */ +#include "OpusDecoderPlugin.h" +#include "OpusHead.hxx" +#include "OpusTags.hxx" +#include "OggUtil.hxx" +#include "OggFind.hxx" +#include "OggSyncState.hxx" +#include "DecoderAPI.hxx" +#include "OggCodec.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "tag/TagBuilder.hxx" +#include "InputStream.hxx" +#include "util/Error.hxx" + +#include <opus.h> +#include <ogg/ogg.h> + +#include <glib.h> + +#include <stdio.h> +#include <string.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "opus" + +static const opus_int32 opus_sample_rate = 48000; + +gcc_pure +static bool +IsOpusHead(const ogg_packet &packet) +{ + return packet.bytes >= 8 && memcmp(packet.packet, "OpusHead", 8) == 0; +} + +gcc_pure +static bool +IsOpusTags(const ogg_packet &packet) +{ + return packet.bytes >= 8 && memcmp(packet.packet, "OpusTags", 8) == 0; +} + +static bool +mpd_opus_init(gcc_unused const config_param ¶m) +{ + g_debug("%s", opus_get_version_string()); + + return true; +} + +class MPDOpusDecoder { + struct decoder *decoder; + struct input_stream *input_stream; + + ogg_stream_state os; + + OpusDecoder *opus_decoder; + opus_int16 *output_buffer; + unsigned output_size; + + bool os_initialized; + bool found_opus; + + int opus_serialno; + + size_t frame_size; + +public: + MPDOpusDecoder(struct decoder *_decoder, + struct input_stream *_input_stream) + :decoder(_decoder), input_stream(_input_stream), + opus_decoder(nullptr), + output_buffer(nullptr), output_size(0), + os_initialized(false), found_opus(false) {} + ~MPDOpusDecoder(); + + bool ReadFirstPage(OggSyncState &oy); + bool ReadNextPage(OggSyncState &oy); + + DecoderCommand HandlePackets(); + DecoderCommand HandlePacket(const ogg_packet &packet); + DecoderCommand HandleBOS(const ogg_packet &packet); + DecoderCommand HandleTags(const ogg_packet &packet); + DecoderCommand HandleAudio(const ogg_packet &packet); +}; + +MPDOpusDecoder::~MPDOpusDecoder() +{ + g_free(output_buffer); + + if (opus_decoder != nullptr) + opus_decoder_destroy(opus_decoder); + + if (os_initialized) + ogg_stream_clear(&os); +} + +inline bool +MPDOpusDecoder::ReadFirstPage(OggSyncState &oy) +{ + assert(!os_initialized); + + if (!oy.ExpectFirstPage(os)) + return false; + + os_initialized = true; + return true; +} + +inline bool +MPDOpusDecoder::ReadNextPage(OggSyncState &oy) +{ + assert(os_initialized); + + ogg_page page; + if (!oy.ExpectPage(page)) + return false; + + const auto page_serialno = ogg_page_serialno(&page); + if (page_serialno != os.serialno) + ogg_stream_reset_serialno(&os, page_serialno); + + ogg_stream_pagein(&os, &page); + return true; +} + +inline DecoderCommand +MPDOpusDecoder::HandlePackets() +{ + ogg_packet packet; + while (ogg_stream_packetout(&os, &packet) == 1) { + auto cmd = HandlePacket(packet); + if (cmd != DecoderCommand::NONE) + return cmd; + } + + return DecoderCommand::NONE; +} + +inline DecoderCommand +MPDOpusDecoder::HandlePacket(const ogg_packet &packet) +{ + if (packet.e_o_s) + return DecoderCommand::STOP; + + if (packet.b_o_s) + return HandleBOS(packet); + else if (!found_opus) + return DecoderCommand::STOP; + + if (IsOpusTags(packet)) + return HandleTags(packet); + + return HandleAudio(packet); +} + +inline DecoderCommand +MPDOpusDecoder::HandleBOS(const ogg_packet &packet) +{ + assert(packet.b_o_s); + + if (found_opus || !IsOpusHead(packet)) + return DecoderCommand::STOP; + + unsigned channels; + if (!ScanOpusHeader(packet.packet, packet.bytes, channels) || + !audio_valid_channel_count(channels)) + return DecoderCommand::STOP; + + assert(opus_decoder == nullptr); + assert(output_buffer == nullptr); + + opus_serialno = os.serialno; + found_opus = true; + + /* TODO: parse attributes from the OpusHead (sample rate, + channels, ...) */ + + int opus_error; + opus_decoder = opus_decoder_create(opus_sample_rate, channels, + &opus_error); + if (opus_decoder == nullptr) { + g_warning("libopus error: %s", + opus_strerror(opus_error)); + return DecoderCommand::STOP; + } + + const AudioFormat audio_format(opus_sample_rate, + SampleFormat::S16, channels); + decoder_initialized(decoder, audio_format, false, -1); + frame_size = audio_format.GetFrameSize(); + + /* allocate an output buffer for 16 bit PCM samples big enough + to hold a quarter second, larger than 120ms required by + libopus */ + output_size = audio_format.sample_rate / 4; + output_buffer = (opus_int16 *) + g_malloc(sizeof(*output_buffer) * output_size * + audio_format.channels); + + return decoder_get_command(decoder); +} + +inline DecoderCommand +MPDOpusDecoder::HandleTags(const ogg_packet &packet) +{ + TagBuilder tag_builder; + + DecoderCommand cmd; + if (ScanOpusTags(packet.packet, packet.bytes, + &add_tag_handler, &tag_builder) && + !tag_builder.IsEmpty()) { + Tag tag; + tag_builder.Commit(tag); + cmd = decoder_tag(decoder, input_stream, std::move(tag)); + } else + cmd = decoder_get_command(decoder); + + return cmd; +} + +inline DecoderCommand +MPDOpusDecoder::HandleAudio(const ogg_packet &packet) +{ + assert(opus_decoder != nullptr); + + int nframes = opus_decode(opus_decoder, + (const unsigned char*)packet.packet, + packet.bytes, + output_buffer, output_size, + 0); + if (nframes < 0) { + g_warning("%s", opus_strerror(nframes)); + return DecoderCommand::STOP; + } + + if (nframes > 0) { + const size_t nbytes = nframes * frame_size; + auto cmd = decoder_data(decoder, input_stream, + output_buffer, nbytes, + 0); + if (cmd != DecoderCommand::NONE) + return cmd; + } + + return DecoderCommand::NONE; +} + +static void +mpd_opus_stream_decode(struct decoder *decoder, + struct input_stream *input_stream) +{ + if (ogg_codec_detect(decoder, input_stream) != OGG_CODEC_OPUS) + return; + + /* rewind the stream, because ogg_codec_detect() has + moved it */ + input_stream->LockSeek(0, SEEK_SET, IgnoreError()); + + MPDOpusDecoder d(decoder, input_stream); + OggSyncState oy(*input_stream, decoder); + + if (!d.ReadFirstPage(oy)) + return; + + while (true) { + auto cmd = d.HandlePackets(); + if (cmd != DecoderCommand::NONE) + break; + + if (!d.ReadNextPage(oy)) + break; + + } +} + +static bool +SeekFindEOS(OggSyncState &oy, ogg_stream_state &os, ogg_packet &packet, + input_stream *is) +{ + if (is->size > 0 && is->size - is->offset < 65536) + return OggFindEOS(oy, os, packet); + + if (!is->CheapSeeking()) + return false; + + oy.Reset(); + + Error error; + return is->LockSeek(-65536, SEEK_END, error) && + oy.ExpectPageSeekIn(os) && + OggFindEOS(oy, os, packet); +} + +static bool +mpd_opus_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + OggSyncState oy(*is); + + ogg_stream_state os; + if (!oy.ExpectFirstPage(os)) + return false; + + /* read at most two more pages */ + unsigned remaining_pages = 2; + + bool result = false; + + ogg_packet packet; + while (true) { + int r = ogg_stream_packetout(&os, &packet); + if (r < 0) { + result = false; + break; + } + + if (r == 0) { + if (remaining_pages-- == 0) + break; + + if (!oy.ExpectPageIn(os)) { + result = false; + break; + } + + continue; + } + + if (packet.b_o_s) { + if (!IsOpusHead(packet)) + break; + + unsigned channels; + if (!ScanOpusHeader(packet.packet, packet.bytes, channels) || + !audio_valid_channel_count(channels)) { + result = false; + break; + } + + result = true; + } else if (!result) + break; + else if (IsOpusTags(packet)) { + if (!ScanOpusTags(packet.packet, packet.bytes, + handler, handler_ctx)) + result = false; + + break; + } + } + + if (packet.e_o_s || SeekFindEOS(oy, os, packet, is)) + tag_handler_invoke_duration(handler, handler_ctx, + packet.granulepos / opus_sample_rate); + + ogg_stream_clear(&os); + + return result; +} + +static const char *const opus_suffixes[] = { + "opus", + "ogg", + "oga", + nullptr +}; + +static const char *const opus_mime_types[] = { + "audio/opus", + nullptr +}; + +const struct decoder_plugin opus_decoder_plugin = { + "opus", + mpd_opus_init, + nullptr, + mpd_opus_stream_decode, + nullptr, + nullptr, + mpd_opus_scan_stream, + nullptr, + opus_suffixes, + opus_mime_types, +}; diff --git a/src/decoder/OpusDecoderPlugin.h b/src/decoder/OpusDecoderPlugin.h new file mode 100644 index 000000000..c95d6ded3 --- /dev/null +++ b/src/decoder/OpusDecoderPlugin.h @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_OPUS_H +#define MPD_DECODER_OPUS_H + +extern const struct decoder_plugin opus_decoder_plugin; + +#endif diff --git a/src/decoder/OpusHead.cxx b/src/decoder/OpusHead.cxx new file mode 100644 index 000000000..c57e08e10 --- /dev/null +++ b/src/decoder/OpusHead.cxx @@ -0,0 +1,44 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "OpusHead.hxx" + +#include <stdint.h> +#include <string.h> + +struct OpusHead { + char signature[8]; + uint8_t version, channels; + uint16_t pre_skip; + uint32_t sample_rate; + uint16_t output_gain; + uint8_t channel_mapping; +}; + +bool +ScanOpusHeader(const void *data, size_t size, unsigned &channels_r) +{ + const OpusHead *h = (const OpusHead *)data; + if (size < 19 || (h->version & 0xf0) != 0) + return false; + + channels_r = h->channels; + return true; +} diff --git a/src/decoder/OpusHead.hxx b/src/decoder/OpusHead.hxx new file mode 100644 index 000000000..9f75c4f70 --- /dev/null +++ b/src/decoder/OpusHead.hxx @@ -0,0 +1,30 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_OPUS_HEAD_HXX +#define MPD_OPUS_HEAD_HXX + +#include "check.h" + +#include <stddef.h> + +bool +ScanOpusHeader(const void *data, size_t size, unsigned &channels_r); + +#endif diff --git a/src/decoder/OpusReader.hxx b/src/decoder/OpusReader.hxx new file mode 100644 index 000000000..7e161fd0f --- /dev/null +++ b/src/decoder/OpusReader.hxx @@ -0,0 +1,100 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_OPUS_READER_HXX +#define MPD_OPUS_READER_HXX + +#include "check.h" + +#include <stdint.h> +#include <string.h> + +class OpusReader { + const uint8_t *p, *const end; + +public: + OpusReader(const void *_p, size_t size) + :p((const uint8_t *)_p), end(p + size) {} + + bool Skip(size_t length) { + p += length; + return p <= end; + } + + const void *Read(size_t length) { + const uint8_t *result = p; + return Skip(length) + ? result + : nullptr; + } + + bool Expect(const void *value, size_t length) { + const void *data = Read(length); + return data != nullptr && memcmp(value, data, length) == 0; + } + + bool ReadByte(uint8_t &value_r) { + if (p >= end) + return false; + + value_r = *p++; + return true; + } + + bool ReadShort(uint16_t &value_r) { + const uint8_t *value = (const uint8_t *)Read(sizeof(value_r)); + if (value == nullptr) + return false; + + value_r = value[0] | (value[1] << 8); + return true; + } + + bool ReadWord(uint32_t &value_r) { + const uint8_t *value = (const uint8_t *)Read(sizeof(value_r)); + if (value == nullptr) + return false; + + value_r = value[0] | (value[1] << 8) + | (value[2] << 16) | (value[3] << 24); + return true; + } + + bool SkipString() { + uint32_t length; + return ReadWord(length) && Skip(length); + } + + char *ReadString() { + uint32_t length; + if (!ReadWord(length)) + return nullptr; + + const char *src = (const char *)Read(length); + if (src == nullptr) + return nullptr; + + char *dest = new char[length + 1]; + memcpy(dest, src, length); + dest[length] = 0; + return dest; + } +}; + +#endif diff --git a/src/decoder/OpusTags.cxx b/src/decoder/OpusTags.cxx new file mode 100644 index 000000000..f09d79c3b --- /dev/null +++ b/src/decoder/OpusTags.cxx @@ -0,0 +1,77 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "OpusTags.hxx" +#include "OpusReader.hxx" +#include "XiphTags.hxx" +#include "tag/TagHandler.hxx" + +#include <stdint.h> +#include <string.h> +#include <stdlib.h> + +static void +ScanOneOpusTag(const char *name, const char *value, + const struct tag_handler *handler, void *ctx) +{ + tag_handler_invoke_pair(handler, ctx, name, value); + + if (handler->tag != nullptr) { + enum tag_type t = tag_table_lookup_i(xiph_tags, name); + if (t != TAG_NUM_OF_ITEM_TYPES) + tag_handler_invoke_tag(handler, ctx, t, value); + } +} + +bool +ScanOpusTags(const void *data, size_t size, + const struct tag_handler *handler, void *ctx) +{ + OpusReader r(data, size); + if (!r.Expect("OpusTags", 8)) + return false; + + if (handler->pair == nullptr && handler->tag == nullptr) + return true; + + if (!r.SkipString()) + return false; + + uint32_t n; + if (!r.ReadWord(n)) + return false; + + while (n-- > 0) { + char *p = r.ReadString(); + if (p == nullptr) + return false; + + char *eq = strchr(p, '='); + if (eq != nullptr && eq > p) { + *eq = 0; + + ScanOneOpusTag(p, eq + 1, handler, ctx); + } + + free(p); + } + + return true; +} diff --git a/src/decoder/OpusTags.hxx b/src/decoder/OpusTags.hxx new file mode 100644 index 000000000..2f3eec844 --- /dev/null +++ b/src/decoder/OpusTags.hxx @@ -0,0 +1,31 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_OPUS_TAGS_HXX +#define MPD_OPUS_TAGS_HXX + +#include "check.h" + +#include <stddef.h> + +bool +ScanOpusTags(const void *data, size_t size, + const struct tag_handler *handler, void *ctx); + +#endif diff --git a/src/decoder/PcmDecoderPlugin.cxx b/src/decoder/PcmDecoderPlugin.cxx new file mode 100644 index 000000000..94867f01d --- /dev/null +++ b/src/decoder/PcmDecoderPlugin.cxx @@ -0,0 +1,119 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "decoder/PcmDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "util/Error.hxx" + +extern "C" { +#include "util/byte_reverse.h" +} + +#include <glib.h> +#include <unistd.h> +#include <string.h> +#include <stdio.h> /* for SEEK_SET */ + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "pcm" + +static void +pcm_stream_decode(struct decoder *decoder, struct input_stream *is) +{ + static constexpr AudioFormat audio_format = { + 44100, + SampleFormat::S16, + 2, + }; + + const char *const mime = is->GetMimeType(); + const bool reverse_endian = mime != nullptr && + strcmp(mime, "audio/x-mpd-cdda-pcm-reverse") == 0; + + const double time_to_size = audio_format.GetTimeToSize(); + + float total_time = -1; + const goffset size = is->GetSize(); + if (size >= 0) + total_time = size / time_to_size; + + decoder_initialized(decoder, audio_format, + is->IsSeekable(), total_time); + + DecoderCommand cmd; + do { + char buffer[4096]; + + size_t nbytes = decoder_read(decoder, is, + buffer, sizeof(buffer)); + + if (nbytes == 0 && is->LockIsEOF()) + break; + + if (reverse_endian) + /* make sure we deliver samples in host byte order */ + reverse_bytes_16((uint16_t *)buffer, + (uint16_t *)buffer, + (uint16_t *)(buffer + nbytes)); + + cmd = nbytes > 0 + ? decoder_data(decoder, is, + buffer, nbytes, 0) + : decoder_get_command(decoder); + if (cmd == DecoderCommand::SEEK) { + goffset offset = (goffset)(time_to_size * + decoder_seek_where(decoder)); + + Error error; + if (is->LockSeek(offset, SEEK_SET, error)) { + decoder_command_finished(decoder); + } else { + g_warning("seeking failed: %s", error.GetMessage()); + decoder_seek_error(decoder); + } + + cmd = DecoderCommand::NONE; + } + } while (cmd == DecoderCommand::NONE); +} + +static const char *const pcm_mime_types[] = { + /* for streams obtained by the cdio_paranoia input plugin */ + "audio/x-mpd-cdda-pcm", + + /* same as above, but with reverse byte order */ + "audio/x-mpd-cdda-pcm-reverse", + + nullptr +}; + +const struct decoder_plugin pcm_decoder_plugin = { + "pcm", + nullptr, + nullptr, + pcm_stream_decode, + nullptr, + nullptr, + nullptr, + nullptr, + nullptr, + pcm_mime_types, +}; diff --git a/src/decoder/PcmDecoderPlugin.hxx b/src/decoder/PcmDecoderPlugin.hxx new file mode 100644 index 000000000..2883e866e --- /dev/null +++ b/src/decoder/PcmDecoderPlugin.hxx @@ -0,0 +1,33 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +/** \file + * + * Not really a decoder; this plugin forwards its input data "as-is". + * + * It was written only to support the "cdio_paranoia" input plugin, + * which does not need a decoder. + */ + +#ifndef MPD_DECODER_PCM_HXX +#define MPD_DECODER_PCM_HXX + +extern const struct decoder_plugin pcm_decoder_plugin; + +#endif diff --git a/src/decoder/SndfileDecoderPlugin.cxx b/src/decoder/SndfileDecoderPlugin.cxx new file mode 100644 index 000000000..56853958c --- /dev/null +++ b/src/decoder/SndfileDecoderPlugin.cxx @@ -0,0 +1,258 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "SndfileDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "util/Error.hxx" + +#include <sndfile.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "sndfile" + +static sf_count_t +sndfile_vio_get_filelen(void *user_data) +{ + const struct input_stream *is = (const struct input_stream *)user_data; + + return is->GetSize(); +} + +static sf_count_t +sndfile_vio_seek(sf_count_t offset, int whence, void *user_data) +{ + struct input_stream *is = (struct input_stream *)user_data; + + if (!is->LockSeek(offset, whence, IgnoreError())) + return -1; + + return is->GetOffset(); +} + +static sf_count_t +sndfile_vio_read(void *ptr, sf_count_t count, void *user_data) +{ + struct input_stream *is = (struct input_stream *)user_data; + + Error error; + size_t nbytes = is->LockRead(ptr, count, error); + if (nbytes == 0 && error.IsDefined()) { + g_warning("%s", error.GetMessage()); + return -1; + } + + return nbytes; +} + +static sf_count_t +sndfile_vio_write(gcc_unused const void *ptr, + gcc_unused sf_count_t count, + gcc_unused void *user_data) +{ + /* no writing! */ + return -1; +} + +static sf_count_t +sndfile_vio_tell(void *user_data) +{ + const struct input_stream *is = (const struct input_stream *)user_data; + + return is->GetOffset(); +} + +/** + * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a + * libsndfile stream. + */ +static SF_VIRTUAL_IO vio = { + sndfile_vio_get_filelen, + sndfile_vio_seek, + sndfile_vio_read, + sndfile_vio_write, + sndfile_vio_tell, +}; + +/** + * Converts a frame number to a timestamp (in seconds). + */ +static float +frame_to_time(sf_count_t frame, const AudioFormat *audio_format) +{ + return (float)frame / (float)audio_format->sample_rate; +} + +/** + * Converts a timestamp (in seconds) to a frame number. + */ +static sf_count_t +time_to_frame(float t, const AudioFormat *audio_format) +{ + return (sf_count_t)(t * audio_format->sample_rate); +} + +static void +sndfile_stream_decode(struct decoder *decoder, struct input_stream *is) +{ + SNDFILE *sf; + SF_INFO info; + size_t frame_size; + sf_count_t read_frames, num_frames; + int buffer[4096]; + + info.format = 0; + + sf = sf_open_virtual(&vio, SFM_READ, &info, is); + if (sf == nullptr) { + g_warning("sf_open_virtual() failed"); + return; + } + + /* for now, always read 32 bit samples. Later, we could lower + MPD's CPU usage by reading 16 bit samples with + sf_readf_short() on low-quality source files. */ + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, info.samplerate, + SampleFormat::S32, + info.channels, error)) { + g_warning("%s", error.GetMessage()); + return; + } + + decoder_initialized(decoder, audio_format, info.seekable, + frame_to_time(info.frames, &audio_format)); + + frame_size = audio_format.GetFrameSize(); + read_frames = sizeof(buffer) / frame_size; + + DecoderCommand cmd; + do { + num_frames = sf_readf_int(sf, buffer, read_frames); + if (num_frames <= 0) + break; + + cmd = decoder_data(decoder, is, + buffer, num_frames * frame_size, + 0); + if (cmd == DecoderCommand::SEEK) { + sf_count_t c = + time_to_frame(decoder_seek_where(decoder), + &audio_format); + c = sf_seek(sf, c, SEEK_SET); + if (c < 0) + decoder_seek_error(decoder); + else + decoder_command_finished(decoder); + cmd = DecoderCommand::NONE; + } + } while (cmd == DecoderCommand::NONE); + + sf_close(sf); +} + +static bool +sndfile_scan_file(const char *path_fs, + const struct tag_handler *handler, void *handler_ctx) +{ + SNDFILE *sf; + SF_INFO info; + const char *p; + + info.format = 0; + + sf = sf_open(path_fs, SFM_READ, &info); + if (sf == nullptr) + return false; + + if (!audio_valid_sample_rate(info.samplerate)) { + sf_close(sf); + g_warning("Invalid sample rate in %s\n", path_fs); + return false; + } + + tag_handler_invoke_duration(handler, handler_ctx, + info.frames / info.samplerate); + + p = sf_get_string(sf, SF_STR_TITLE); + if (p != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_TITLE, p); + + p = sf_get_string(sf, SF_STR_ARTIST); + if (p != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_ARTIST, p); + + p = sf_get_string(sf, SF_STR_DATE); + if (p != nullptr) + tag_handler_invoke_tag(handler, handler_ctx, + TAG_DATE, p); + + sf_close(sf); + + return true; +} + +static const char *const sndfile_suffixes[] = { + "wav", "aiff", "aif", /* Microsoft / SGI / Apple */ + "au", "snd", /* Sun / DEC / NeXT */ + "paf", /* Paris Audio File */ + "iff", "svx", /* Commodore Amiga IFF / SVX */ + "sf", /* IRCAM */ + "voc", /* Creative */ + "w64", /* Soundforge */ + "pvf", /* Portable Voice Format */ + "xi", /* Fasttracker */ + "htk", /* HMM Tool Kit */ + "caf", /* Apple */ + "sd2", /* Sound Designer II */ + + /* libsndfile also supports FLAC and Ogg Vorbis, but only by + linking with libFLAC and libvorbis - we can do better, we + have native plugins for these libraries */ + + nullptr +}; + +static const char *const sndfile_mime_types[] = { + "audio/x-wav", + "audio/x-aiff", + + /* what are the MIME types of the other supported formats? */ + + nullptr +}; + +const struct decoder_plugin sndfile_decoder_plugin = { + "sndfile", + nullptr, + nullptr, + sndfile_stream_decode, + nullptr, + sndfile_scan_file, + nullptr, + nullptr, + sndfile_suffixes, + sndfile_mime_types, +}; diff --git a/src/decoder/SndfileDecoderPlugin.hxx b/src/decoder/SndfileDecoderPlugin.hxx new file mode 100644 index 000000000..ba60fafd0 --- /dev/null +++ b/src/decoder/SndfileDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_SNDFILE_HXX +#define MPD_DECODER_SNDFILE_HXX + +extern const struct decoder_plugin sndfile_decoder_plugin; + +#endif diff --git a/src/decoder/VorbisComments.cxx b/src/decoder/VorbisComments.cxx new file mode 100644 index 000000000..c8eeb09cd --- /dev/null +++ b/src/decoder/VorbisComments.cxx @@ -0,0 +1,147 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "VorbisComments.hxx" +#include "XiphTags.hxx" +#include "tag/Tag.hxx" +#include "tag/TagTable.hxx" +#include "tag/TagHandler.hxx" +#include "tag/TagBuilder.hxx" +#include "replay_gain_info.h" + +#include <glib.h> +#include <assert.h> +#include <stddef.h> +#include <string.h> +#include <stdlib.h> + +static const char * +vorbis_comment_value(const char *comment, const char *needle) +{ + size_t len = strlen(needle); + + if (g_ascii_strncasecmp(comment, needle, len) == 0 && + comment[len] == '=') + return comment + len + 1; + + return NULL; +} + +bool +vorbis_comments_to_replay_gain(struct replay_gain_info *rgi, char **comments) +{ + const char *temp; + bool found = false; + + replay_gain_info_init(rgi); + + while (*comments) { + if ((temp = + vorbis_comment_value(*comments, "replaygain_track_gain"))) { + rgi->tuples[REPLAY_GAIN_TRACK].gain = atof(temp); + found = true; + } else if ((temp = vorbis_comment_value(*comments, + "replaygain_album_gain"))) { + rgi->tuples[REPLAY_GAIN_ALBUM].gain = atof(temp); + found = true; + } else if ((temp = vorbis_comment_value(*comments, + "replaygain_track_peak"))) { + rgi->tuples[REPLAY_GAIN_TRACK].peak = atof(temp); + found = true; + } else if ((temp = vorbis_comment_value(*comments, + "replaygain_album_peak"))) { + rgi->tuples[REPLAY_GAIN_ALBUM].peak = atof(temp); + found = true; + } + + comments++; + } + + return found; +} + +/** + * Check if the comment's name equals the passed name, and if so, copy + * the comment value into the tag. + */ +static bool +vorbis_copy_comment(const char *comment, + const char *name, enum tag_type tag_type, + const struct tag_handler *handler, void *handler_ctx) +{ + const char *value; + + value = vorbis_comment_value(comment, name); + if (value != NULL) { + tag_handler_invoke_tag(handler, handler_ctx, tag_type, value); + return true; + } + + return false; +} + +static void +vorbis_scan_comment(const char *comment, + const struct tag_handler *handler, void *handler_ctx) +{ + if (handler->pair != NULL) { + char *name = g_strdup((const char*)comment); + char *value = strchr(name, '='); + + if (value != NULL && value > name) { + *value++ = 0; + tag_handler_invoke_pair(handler, handler_ctx, + name, value); + } + + g_free(name); + } + + for (const struct tag_table *i = xiph_tags; i->name != NULL; ++i) + if (vorbis_copy_comment(comment, i->name, i->type, + handler, handler_ctx)) + return; + + for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) + if (vorbis_copy_comment(comment, + tag_item_names[i], tag_type(i), + handler, handler_ctx)) + return; +} + +void +vorbis_comments_scan(char **comments, + const struct tag_handler *handler, void *handler_ctx) +{ + while (*comments) + vorbis_scan_comment(*comments++, + handler, handler_ctx); + +} + +Tag * +vorbis_comments_to_tag(char **comments) +{ + TagBuilder tag_builder; + vorbis_comments_scan(comments, &add_tag_handler, &tag_builder); + return tag_builder.IsEmpty() + ? nullptr + : tag_builder.Commit(); +} diff --git a/src/decoder/VorbisComments.hxx b/src/decoder/VorbisComments.hxx new file mode 100644 index 000000000..7a8374785 --- /dev/null +++ b/src/decoder/VorbisComments.hxx @@ -0,0 +1,39 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_VORBIS_COMMENTS_HXX +#define MPD_VORBIS_COMMENTS_HXX + +#include "check.h" + +struct replay_gain_info; +struct tag_handler; +struct Tag; + +bool +vorbis_comments_to_replay_gain(struct replay_gain_info *rgi, char **comments); + +void +vorbis_comments_scan(char **comments, + const struct tag_handler *handler, void *handler_ctx); + +Tag * +vorbis_comments_to_tag(char **comments); + +#endif diff --git a/src/decoder/VorbisDecoderPlugin.cxx b/src/decoder/VorbisDecoderPlugin.cxx new file mode 100644 index 000000000..a4a938aa8 --- /dev/null +++ b/src/decoder/VorbisDecoderPlugin.cxx @@ -0,0 +1,356 @@ +/* + * Copyright (C) 2003-2011 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "VorbisDecoderPlugin.h" +#include "VorbisComments.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "OggCodec.hxx" +#include "util/Error.hxx" +#include "util/UriUtil.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" + +#ifndef HAVE_TREMOR +#define OV_EXCLUDE_STATIC_CALLBACKS +#include <vorbis/vorbisfile.h> +#else +#include <tremor/ivorbisfile.h> +/* Macros to make Tremor's API look like libogg. Tremor always + returns host-byte-order 16-bit signed data, and uses integer + milliseconds where libogg uses double seconds. +*/ +#define ov_read(VF, BUFFER, LENGTH, BIGENDIANP, WORD, SGNED, BITSTREAM) \ + ov_read(VF, BUFFER, LENGTH, BITSTREAM) +#define ov_time_total(VF, I) ((double)ov_time_total(VF, I)/1000) +#define ov_time_tell(VF) ((double)ov_time_tell(VF)/1000) +#define ov_time_seek_page(VF, S) (ov_time_seek_page(VF, (S)*1000)) +#endif /* HAVE_TREMOR */ + +#include <glib.h> + +#include <assert.h> +#include <errno.h> +#include <unistd.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "vorbis" + +#if G_BYTE_ORDER == G_BIG_ENDIAN +#define VORBIS_BIG_ENDIAN true +#else +#define VORBIS_BIG_ENDIAN false +#endif + +struct vorbis_input_stream { + struct decoder *decoder; + + struct input_stream *input_stream; + bool seekable; +}; + +static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *data) +{ + struct vorbis_input_stream *vis = (struct vorbis_input_stream *)data; + size_t ret = decoder_read(vis->decoder, vis->input_stream, + ptr, size * nmemb); + + errno = 0; + + return ret / size; +} + +static int ogg_seek_cb(void *data, ogg_int64_t offset, int whence) +{ + struct vorbis_input_stream *vis = (struct vorbis_input_stream *)data; + + Error error; + return vis->seekable && + (!vis->decoder || decoder_get_command(vis->decoder) != DecoderCommand::STOP) && + vis->input_stream->LockSeek(offset, whence, error) + ? 0 : -1; +} + +/* TODO: check Ogg libraries API and see if we can just not have this func */ +static int ogg_close_cb(gcc_unused void *data) +{ + return 0; +} + +static long ogg_tell_cb(void *data) +{ + struct vorbis_input_stream *vis = (struct vorbis_input_stream *)data; + + return (long)vis->input_stream->offset; +} + +static const ov_callbacks vorbis_is_callbacks = { + ogg_read_cb, + ogg_seek_cb, + ogg_close_cb, + ogg_tell_cb, +}; + +static const char * +vorbis_strerror(int code) +{ + switch (code) { + case OV_EREAD: + return "read error"; + + case OV_ENOTVORBIS: + return "not vorbis stream"; + + case OV_EVERSION: + return "vorbis version mismatch"; + + case OV_EBADHEADER: + return "invalid vorbis header"; + + case OV_EFAULT: + return "internal logic error"; + + default: + return "unknown error"; + } +} + +static bool +vorbis_is_open(struct vorbis_input_stream *vis, OggVorbis_File *vf, + struct decoder *decoder, struct input_stream *input_stream) +{ + vis->decoder = decoder; + vis->input_stream = input_stream; + vis->seekable = input_stream->CheapSeeking(); + + int ret = ov_open_callbacks(vis, vf, NULL, 0, vorbis_is_callbacks); + if (ret < 0) { + if (decoder == NULL || + decoder_get_command(decoder) == DecoderCommand::NONE) + g_warning("Failed to open Ogg Vorbis stream: %s", + vorbis_strerror(ret)); + return false; + } + + return true; +} + +static void +vorbis_send_comments(struct decoder *decoder, struct input_stream *is, + char **comments) +{ + Tag *tag = vorbis_comments_to_tag(comments); + if (!tag) + return; + + decoder_tag(decoder, is, std::move(*tag)); + delete tag; +} + +#ifndef HAVE_TREMOR +static void +vorbis_interleave(float *dest, const float *const*src, + unsigned nframes, unsigned channels) +{ + for (const float *const*src_end = src + channels; + src != src_end; ++src, ++dest) { + float *d = dest; + for (const float *s = *src, *s_end = s + nframes; + s != s_end; ++s, d += channels) + *d = *s; + } +} +#endif + +/* public */ +static void +vorbis_stream_decode(struct decoder *decoder, + struct input_stream *input_stream) +{ + if (ogg_codec_detect(decoder, input_stream) != OGG_CODEC_VORBIS) + return; + + /* rewind the stream, because ogg_codec_detect() has + moved it */ + input_stream->LockSeek(0, SEEK_SET, IgnoreError()); + + struct vorbis_input_stream vis; + OggVorbis_File vf; + if (!vorbis_is_open(&vis, &vf, decoder, input_stream)) + return; + + const vorbis_info *vi = ov_info(&vf, -1); + if (vi == NULL) { + g_warning("ov_info() has failed"); + return; + } + + Error error; + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, vi->rate, +#ifdef HAVE_TREMOR + SampleFormat::S16, +#else + SampleFormat::FLOAT, +#endif + vi->channels, error)) { + g_warning("%s", error.GetMessage()); + return; + } + + float total_time = ov_time_total(&vf, -1); + if (total_time < 0) + total_time = 0; + + decoder_initialized(decoder, audio_format, vis.seekable, total_time); + +#ifdef HAVE_TREMOR + char buffer[4096]; +#else + float buffer[2048]; + const int frames_per_buffer = + G_N_ELEMENTS(buffer) / audio_format.channels; + const unsigned frame_size = sizeof(buffer[0]) * audio_format.channels; +#endif + + int prev_section = -1; + unsigned kbit_rate = 0; + + DecoderCommand cmd = decoder_get_command(decoder); + do { + if (cmd == DecoderCommand::SEEK) { + double seek_where = decoder_seek_where(decoder); + if (0 == ov_time_seek_page(&vf, seek_where)) { + decoder_command_finished(decoder); + } else + decoder_seek_error(decoder); + } + + int current_section; + +#ifdef HAVE_TREMOR + long nbytes = ov_read(&vf, buffer, sizeof(buffer), + VORBIS_BIG_ENDIAN, 2, 1, + ¤t_section); +#else + float **per_channel; + long nframes = ov_read_float(&vf, &per_channel, + frames_per_buffer, + ¤t_section); + long nbytes = nframes; + if (nframes > 0) { + vorbis_interleave(buffer, + (const float*const*)per_channel, + nframes, audio_format.channels); + nbytes *= frame_size; + } +#endif + + if (nbytes == OV_HOLE) /* bad packet */ + nbytes = 0; + else if (nbytes <= 0) + /* break on EOF or other error */ + break; + + if (current_section != prev_section) { + vi = ov_info(&vf, -1); + if (vi == NULL) { + g_warning("ov_info() has failed"); + break; + } + + if (vi->rate != (long)audio_format.sample_rate || + vi->channels != (int)audio_format.channels) { + /* we don't support audio format + change yet */ + g_warning("audio format change, stopping here"); + break; + } + + char **comments = ov_comment(&vf, -1)->user_comments; + vorbis_send_comments(decoder, input_stream, comments); + + struct replay_gain_info rgi; + if (vorbis_comments_to_replay_gain(&rgi, comments)) + decoder_replay_gain(decoder, &rgi); + + prev_section = current_section; + } + + long test = ov_bitrate_instant(&vf); + if (test > 0) + kbit_rate = test / 1000; + + cmd = decoder_data(decoder, input_stream, + buffer, nbytes, + kbit_rate); + } while (cmd != DecoderCommand::STOP); + + ov_clear(&vf); +} + +static bool +vorbis_scan_stream(struct input_stream *is, + const struct tag_handler *handler, void *handler_ctx) +{ + struct vorbis_input_stream vis; + OggVorbis_File vf; + + if (!vorbis_is_open(&vis, &vf, NULL, is)) + return false; + + tag_handler_invoke_duration(handler, handler_ctx, + (int)(ov_time_total(&vf, -1) + 0.5)); + + vorbis_comments_scan(ov_comment(&vf, -1)->user_comments, + handler, handler_ctx); + + ov_clear(&vf); + return true; +} + +static const char *const vorbis_suffixes[] = { + "ogg", "oga", NULL +}; + +static const char *const vorbis_mime_types[] = { + "application/ogg", + "application/x-ogg", + "audio/ogg", + "audio/vorbis", + "audio/vorbis+ogg", + "audio/x-ogg", + "audio/x-vorbis", + "audio/x-vorbis+ogg", + NULL +}; + +const struct decoder_plugin vorbis_decoder_plugin = { + "vorbis", + nullptr, + nullptr, + vorbis_stream_decode, + nullptr, + nullptr, + vorbis_scan_stream, + nullptr, + vorbis_suffixes, + vorbis_mime_types +}; diff --git a/src/decoder/VorbisDecoderPlugin.h b/src/decoder/VorbisDecoderPlugin.h new file mode 100644 index 000000000..618c9ffde --- /dev/null +++ b/src/decoder/VorbisDecoderPlugin.h @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2012 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_VORBIS_H +#define MPD_DECODER_VORBIS_H + +extern const struct decoder_plugin vorbis_decoder_plugin; + +#endif diff --git a/src/decoder/WavpackDecoderPlugin.cxx b/src/decoder/WavpackDecoderPlugin.cxx new file mode 100644 index 000000000..ecabafefe --- /dev/null +++ b/src/decoder/WavpackDecoderPlugin.cxx @@ -0,0 +1,598 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "WavpackDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "InputStream.hxx" +#include "CheckAudioFormat.hxx" +#include "tag/TagHandler.hxx" +#include "tag/ApeTag.hxx" +#include "util/Error.hxx" + +#include <wavpack/wavpack.h> +#include <glib.h> + +#include <assert.h> +#include <stdio.h> +#include <stdlib.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "wavpack" + +#define ERRORLEN 80 + +/** A pointer type for format converter function. */ +typedef void (*format_samples_t)( + int bytes_per_sample, + void *buffer, uint32_t count +); + +/* + * This function has been borrowed from the tiny player found on + * wavpack.com. Modifications were required because mpd only handles + * max 24-bit samples. + */ +static void +format_samples_int(int bytes_per_sample, void *buffer, uint32_t count) +{ + int32_t *src = (int32_t *)buffer; + + switch (bytes_per_sample) { + case 1: { + int8_t *dst = (int8_t *)buffer; + /* + * The asserts like the following one are because we do the + * formatting of samples within a single buffer. The size + * of the output samples never can be greater than the size + * of the input ones. Otherwise we would have an overflow. + */ + static_assert(sizeof(*dst) <= sizeof(*src), "Wrong size"); + + /* pass through and align 8-bit samples */ + while (count--) { + *dst++ = *src++; + } + break; + } + case 2: { + uint16_t *dst = (uint16_t *)buffer; + static_assert(sizeof(*dst) <= sizeof(*src), "Wrong size"); + + /* pass through and align 16-bit samples */ + while (count--) { + *dst++ = *src++; + } + break; + } + + case 3: + case 4: + /* do nothing */ + break; + } +} + +/* + * This function converts floating point sample data to 24-bit integer. + */ +static void +format_samples_float(gcc_unused int bytes_per_sample, void *buffer, + uint32_t count) +{ + float *p = (float *)buffer; + + while (count--) { + *p /= (1 << 23); + ++p; + } +} + +/** + * Choose a MPD sample format from libwavpacks' number of bits. + */ +static SampleFormat +wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample) +{ + if (is_float) + return SampleFormat::FLOAT; + + switch (bytes_per_sample) { + case 1: + return SampleFormat::S8; + + case 2: + return SampleFormat::S16; + + case 3: + return SampleFormat::S24_P32; + + case 4: + return SampleFormat::S32; + + default: + return SampleFormat::UNDEFINED; + } +} + +/* + * This does the main decoding thing. + * Requires an already opened WavpackContext. + */ +static void +wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) +{ + bool is_float; + SampleFormat sample_format; + AudioFormat audio_format; + format_samples_t format_samples; + float total_time; + int bytes_per_sample, output_sample_size; + + is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0; + sample_format = + wavpack_bits_to_sample_format(is_float, + WavpackGetBytesPerSample(wpc)); + + Error error; + if (!audio_format_init_checked(audio_format, + WavpackGetSampleRate(wpc), + sample_format, + WavpackGetNumChannels(wpc), error)) { + g_warning("%s", error.GetMessage()); + return; + } + + if (is_float) { + format_samples = format_samples_float; + } else { + format_samples = format_samples_int; + } + + total_time = WavpackGetNumSamples(wpc); + total_time /= audio_format.sample_rate; + bytes_per_sample = WavpackGetBytesPerSample(wpc); + output_sample_size = audio_format.GetFrameSize(); + + /* wavpack gives us all kind of samples in a 32-bit space */ + int32_t chunk[1024]; + const uint32_t samples_requested = G_N_ELEMENTS(chunk) / + audio_format.channels; + + decoder_initialized(decoder, audio_format, can_seek, total_time); + + DecoderCommand cmd = decoder_get_command(decoder); + while (cmd != DecoderCommand::STOP) { + if (cmd == DecoderCommand::SEEK) { + if (can_seek) { + unsigned where = decoder_seek_where(decoder) * + audio_format.sample_rate; + + if (WavpackSeekSample(wpc, where)) { + decoder_command_finished(decoder); + } else { + decoder_seek_error(decoder); + } + } else { + decoder_seek_error(decoder); + } + } + + uint32_t samples_got = WavpackUnpackSamples(wpc, chunk, + samples_requested); + if (samples_got == 0) + break; + + int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 + + 0.5); + format_samples(bytes_per_sample, chunk, + samples_got * audio_format.channels); + + cmd = decoder_data(decoder, NULL, chunk, + samples_got * output_sample_size, + bitrate); + } +} + +/** + * Locate and parse a floating point tag. Returns true if it was + * found. + */ +static bool +wavpack_tag_float(WavpackContext *wpc, const char *key, float *value_r) +{ + char buffer[64]; + int ret; + + ret = WavpackGetTagItem(wpc, key, buffer, sizeof(buffer)); + if (ret <= 0) + return false; + + *value_r = atof(buffer); + return true; +} + +static bool +wavpack_replaygain(struct replay_gain_info *replay_gain_info, + WavpackContext *wpc) +{ + bool found = false; + + replay_gain_info_init(replay_gain_info); + + found |= wavpack_tag_float( + wpc, "replaygain_track_gain", + &replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain + ); + found |= wavpack_tag_float( + wpc, "replaygain_track_peak", + &replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak + ); + found |= wavpack_tag_float( + wpc, "replaygain_album_gain", + &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain + ); + found |= wavpack_tag_float( + wpc, "replaygain_album_peak", + &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak + ); + + return found; +} + +static void +wavpack_scan_tag_item(WavpackContext *wpc, const char *name, + enum tag_type type, + const struct tag_handler *handler, void *handler_ctx) +{ + char buffer[1024]; + int len = WavpackGetTagItem(wpc, name, buffer, sizeof(buffer)); + if (len <= 0 || (unsigned)len >= sizeof(buffer)) + return; + + tag_handler_invoke_tag(handler, handler_ctx, type, buffer); + +} + +static void +wavpack_scan_pair(WavpackContext *wpc, const char *name, + const struct tag_handler *handler, void *handler_ctx) +{ + char buffer[8192]; + int len = WavpackGetTagItem(wpc, name, buffer, sizeof(buffer)); + if (len <= 0 || (unsigned)len >= sizeof(buffer)) + return; + + tag_handler_invoke_pair(handler, handler_ctx, name, buffer); +} + +/* + * Reads metainfo from the specified file. + */ +static bool +wavpack_scan_file(const char *fname, + const struct tag_handler *handler, void *handler_ctx) +{ + WavpackContext *wpc; + char error[ERRORLEN]; + + wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0); + if (wpc == NULL) { + g_warning( + "failed to open WavPack file \"%s\": %s\n", + fname, error + ); + return false; + } + + tag_handler_invoke_duration(handler, handler_ctx, + WavpackGetNumSamples(wpc) / + WavpackGetSampleRate(wpc)); + + /* the WavPack format implies APEv2 tags, which means we can + reuse the mapping from tag_ape.c */ + + for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) { + const char *name = tag_item_names[i]; + if (name != NULL) + wavpack_scan_tag_item(wpc, name, (enum tag_type)i, + handler, handler_ctx); + } + + for (const struct tag_table *i = ape_tags; i->name != NULL; ++i) + wavpack_scan_tag_item(wpc, i->name, i->type, + handler, handler_ctx); + + if (handler->pair != NULL) { + char name[64]; + + for (int i = 0, n = WavpackGetNumTagItems(wpc); + i < n; ++i) { + int len = WavpackGetTagItemIndexed(wpc, i, name, + sizeof(name)); + if (len <= 0 || (unsigned)len >= sizeof(name)) + continue; + + wavpack_scan_pair(wpc, name, handler, handler_ctx); + } + } + + WavpackCloseFile(wpc); + + return true; +} + +/* + * mpd input_stream <=> WavpackStreamReader wrapper callbacks + */ + +/* This struct is needed for per-stream last_byte storage. */ +struct wavpack_input { + struct decoder *decoder; + struct input_stream *is; + /* Needed for push_back_byte() */ + int last_byte; +}; + +/** + * Little wrapper for struct wavpack_input to cast from void *. + */ +static struct wavpack_input * +wpin(void *id) +{ + assert(id); + return (struct wavpack_input *)id; +} + +static int32_t +wavpack_input_read_bytes(void *id, void *data, int32_t bcount) +{ + uint8_t *buf = (uint8_t *)data; + int32_t i = 0; + + if (wpin(id)->last_byte != EOF) { + *buf++ = wpin(id)->last_byte; + wpin(id)->last_byte = EOF; + --bcount; + ++i; + } + + /* wavpack fails if we return a partial read, so we just wait + until the buffer is full */ + while (bcount > 0) { + size_t nbytes = decoder_read( + wpin(id)->decoder, wpin(id)->is, buf, bcount + ); + if (nbytes == 0) { + /* EOF, error or a decoder command */ + break; + } + + i += nbytes; + bcount -= nbytes; + buf += nbytes; + } + + return i; +} + +static uint32_t +wavpack_input_get_pos(void *id) +{ + return wpin(id)->is->offset; +} + +static int +wavpack_input_set_pos_abs(void *id, uint32_t pos) +{ + return wpin(id)->is->LockSeek(pos, SEEK_SET, IgnoreError()) ? 0 : -1; +} + +static int +wavpack_input_set_pos_rel(void *id, int32_t delta, int mode) +{ + return wpin(id)->is->LockSeek(delta, mode, IgnoreError()) ? 0 : -1; +} + +static int +wavpack_input_push_back_byte(void *id, int c) +{ + if (wpin(id)->last_byte == EOF) { + wpin(id)->last_byte = c; + return c; + } else { + return EOF; + } +} + +static uint32_t +wavpack_input_get_length(void *id) +{ + if (wpin(id)->is->size < 0) + return 0; + + return wpin(id)->is->size; +} + +static int +wavpack_input_can_seek(void *id) +{ + return wpin(id)->is->seekable; +} + +static WavpackStreamReader mpd_is_reader = { + wavpack_input_read_bytes, + wavpack_input_get_pos, + wavpack_input_set_pos_abs, + wavpack_input_set_pos_rel, + wavpack_input_push_back_byte, + wavpack_input_get_length, + wavpack_input_can_seek, + nullptr /* no need to write edited tags */ +}; + +static void +wavpack_input_init(struct wavpack_input *isp, struct decoder *decoder, + struct input_stream *is) +{ + isp->decoder = decoder; + isp->is = is; + isp->last_byte = EOF; +} + +static struct input_stream * +wavpack_open_wvc(struct decoder *decoder, const char *uri, + Mutex &mutex, Cond &cond, + struct wavpack_input *wpi) +{ + struct input_stream *is_wvc; + char *wvc_url = NULL; + char first_byte; + size_t nbytes; + + /* + * As we use dc->utf8url, this function will be bad for + * single files. utf8url is not absolute file path :/ + */ + if (uri == NULL) + return nullptr; + + wvc_url = g_strconcat(uri, "c", NULL); + + is_wvc = input_stream::Open(wvc_url, mutex, cond, IgnoreError()); + g_free(wvc_url); + + if (is_wvc == NULL) + return NULL; + + /* + * And we try to buffer in order to get know + * about a possible 404 error. + */ + nbytes = decoder_read( + decoder, is_wvc, &first_byte, sizeof(first_byte) + ); + if (nbytes == 0) { + is_wvc->Close(); + return NULL; + } + + /* push it back */ + wavpack_input_init(wpi, decoder, is_wvc); + wpi->last_byte = first_byte; + return is_wvc; +} + +/* + * Decodes a stream. + */ +static void +wavpack_streamdecode(struct decoder * decoder, struct input_stream *is) +{ + char error[ERRORLEN]; + WavpackContext *wpc; + struct input_stream *is_wvc; + int open_flags = OPEN_NORMALIZE; + struct wavpack_input isp, isp_wvc; + bool can_seek = is->seekable; + + is_wvc = wavpack_open_wvc(decoder, is->uri.c_str(), + is->mutex, is->cond, + &isp_wvc); + if (is_wvc != NULL) { + open_flags |= OPEN_WVC; + can_seek &= is_wvc->seekable; + } + + if (!can_seek) { + open_flags |= OPEN_STREAMING; + } + + wavpack_input_init(&isp, decoder, is); + wpc = WavpackOpenFileInputEx( + &mpd_is_reader, &isp, + open_flags & OPEN_WVC ? &isp_wvc : NULL, + error, open_flags, 23 + ); + + if (wpc == NULL) { + g_warning("failed to open WavPack stream: %s\n", error); + return; + } + + wavpack_decode(decoder, wpc, can_seek); + + WavpackCloseFile(wpc); + if (open_flags & OPEN_WVC) { + is_wvc->Close(); + } +} + +/* + * Decodes a file. + */ +static void +wavpack_filedecode(struct decoder *decoder, const char *fname) +{ + char error[ERRORLEN]; + WavpackContext *wpc; + + wpc = WavpackOpenFileInput( + fname, error, + OPEN_TAGS | OPEN_WVC | OPEN_NORMALIZE, 23 + ); + if (wpc == NULL) { + g_warning( + "failed to open WavPack file \"%s\": %s\n", + fname, error + ); + return; + } + + struct replay_gain_info replay_gain_info; + if (wavpack_replaygain(&replay_gain_info, wpc)) + decoder_replay_gain(decoder, &replay_gain_info); + + wavpack_decode(decoder, wpc, true); + + WavpackCloseFile(wpc); +} + +static char const *const wavpack_suffixes[] = { + "wv", + NULL +}; + +static char const *const wavpack_mime_types[] = { + "audio/x-wavpack", + NULL +}; + +const struct decoder_plugin wavpack_decoder_plugin = { + "wavpack", + nullptr, + nullptr, + wavpack_streamdecode, + wavpack_filedecode, + wavpack_scan_file, + nullptr, + nullptr, + wavpack_suffixes, + wavpack_mime_types +}; diff --git a/src/decoder/WavpackDecoderPlugin.hxx b/src/decoder/WavpackDecoderPlugin.hxx new file mode 100644 index 000000000..9ebe6354f --- /dev/null +++ b/src/decoder/WavpackDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_WAVPACK_HXX +#define MPD_DECODER_WAVPACK_HXX + +extern const struct decoder_plugin wavpack_decoder_plugin; + +#endif diff --git a/src/decoder/WildmidiDecoderPlugin.cxx b/src/decoder/WildmidiDecoderPlugin.cxx new file mode 100644 index 000000000..3a057ca2c --- /dev/null +++ b/src/decoder/WildmidiDecoderPlugin.cxx @@ -0,0 +1,160 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "WildmidiDecoderPlugin.hxx" +#include "DecoderAPI.hxx" +#include "tag/TagHandler.hxx" +#include "util/Error.hxx" +#include "fs/Path.hxx" +#include "fs/FileSystem.hxx" +#include "system/FatalError.hxx" + +#include <glib.h> + +extern "C" { +#include <wildmidi_lib.h> +} + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "wildmidi" + +static constexpr unsigned WILDMIDI_SAMPLE_RATE = 48000; + +static bool +wildmidi_init(const config_param ¶m) +{ + Error error; + const Path path = param.GetBlockPath("config_file", + "/etc/timidity/timidity.cfg", + error); + if (path.IsNull()) + FatalError(error); + + if (!FileExists(path)) { + const auto utf8 = path.ToUTF8(); + g_debug("configuration file does not exist: %s", utf8.c_str()); + return false; + } + + return WildMidi_Init(path.c_str(), WILDMIDI_SAMPLE_RATE, 0) == 0; +} + +static void +wildmidi_finish(void) +{ + WildMidi_Shutdown(); +} + +static void +wildmidi_file_decode(struct decoder *decoder, const char *path_fs) +{ + static constexpr AudioFormat audio_format = { + WILDMIDI_SAMPLE_RATE, + SampleFormat::S16, + 2, + }; + midi *wm; + const struct _WM_Info *info; + + wm = WildMidi_Open(path_fs); + if (wm == nullptr) + return; + + info = WildMidi_GetInfo(wm); + if (info == nullptr) { + WildMidi_Close(wm); + return; + } + + decoder_initialized(decoder, audio_format, true, + info->approx_total_samples / WILDMIDI_SAMPLE_RATE); + + DecoderCommand cmd; + do { + char buffer[4096]; + int len; + + info = WildMidi_GetInfo(wm); + if (info == nullptr) + break; + + len = WildMidi_GetOutput(wm, buffer, sizeof(buffer)); + if (len <= 0) + break; + + cmd = decoder_data(decoder, nullptr, buffer, len, 0); + + if (cmd == DecoderCommand::SEEK) { + unsigned long seek_where = WILDMIDI_SAMPLE_RATE * + decoder_seek_where(decoder); + +#ifdef HAVE_WILDMIDI_SAMPLED_SEEK + WildMidi_SampledSeek(wm, &seek_where); +#else + WildMidi_FastSeek(wm, &seek_where); +#endif + decoder_command_finished(decoder); + cmd = DecoderCommand::NONE; + } + + } while (cmd == DecoderCommand::NONE); + + WildMidi_Close(wm); +} + +static bool +wildmidi_scan_file(const char *path_fs, + const struct tag_handler *handler, void *handler_ctx) +{ + midi *wm = WildMidi_Open(path_fs); + if (wm == nullptr) + return false; + + const struct _WM_Info *info = WildMidi_GetInfo(wm); + if (info == nullptr) { + WildMidi_Close(wm); + return false; + } + + int duration = info->approx_total_samples / WILDMIDI_SAMPLE_RATE; + tag_handler_invoke_duration(handler, handler_ctx, duration); + + WildMidi_Close(wm); + + return true; +} + +static const char *const wildmidi_suffixes[] = { + "mid", + nullptr +}; + +const struct decoder_plugin wildmidi_decoder_plugin = { + "wildmidi", + wildmidi_init, + wildmidi_finish, + nullptr, + wildmidi_file_decode, + wildmidi_scan_file, + nullptr, + nullptr, + wildmidi_suffixes, + nullptr, +}; diff --git a/src/decoder/WildmidiDecoderPlugin.hxx b/src/decoder/WildmidiDecoderPlugin.hxx new file mode 100644 index 000000000..956b72299 --- /dev/null +++ b/src/decoder/WildmidiDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_DECODER_WILDMIDI_HXX +#define MPD_DECODER_WILDMIDI_HXX + +extern const struct decoder_plugin wildmidi_decoder_plugin; + +#endif diff --git a/src/decoder/XiphTags.cxx b/src/decoder/XiphTags.cxx new file mode 100644 index 000000000..b9958a19a --- /dev/null +++ b/src/decoder/XiphTags.cxx @@ -0,0 +1,28 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "XiphTags.hxx" + +const struct tag_table xiph_tags[] = { + { "tracknumber", TAG_TRACK }, + { "discnumber", TAG_DISC }, + { "album artist", TAG_ALBUM_ARTIST }, + { nullptr, TAG_NUM_OF_ITEM_TYPES } +}; diff --git a/src/decoder/XiphTags.hxx b/src/decoder/XiphTags.hxx new file mode 100644 index 000000000..606dfef10 --- /dev/null +++ b/src/decoder/XiphTags.hxx @@ -0,0 +1,28 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_XIPH_TAGS_HXX +#define MPD_XIPH_TAGS_HXX + +#include "check.h" +#include "tag/TagTable.hxx" + +extern const struct tag_table xiph_tags[]; + +#endif diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c deleted file mode 100644 index d7f0c4a8a..000000000 --- a/src/decoder/_flac_common.c +++ /dev/null @@ -1,228 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* - * Common data structures and functions used by FLAC and OggFLAC - */ - -#include "config.h" -#include "_flac_common.h" -#include "flac_metadata.h" -#include "flac_pcm.h" -#include "audio_check.h" - -#include <glib.h> - -#include <assert.h> - -void -flac_data_init(struct flac_data *data, struct decoder * decoder, - struct input_stream *input_stream) -{ - pcm_buffer_init(&data->buffer); - - data->unsupported = false; - data->initialized = false; - data->total_frames = 0; - data->first_frame = 0; - data->next_frame = 0; - - data->position = 0; - data->decoder = decoder; - data->input_stream = input_stream; - data->tag = NULL; -} - -void -flac_data_deinit(struct flac_data *data) -{ - pcm_buffer_deinit(&data->buffer); - - if (data->tag != NULL) - tag_free(data->tag); -} - -static enum sample_format -flac_sample_format(unsigned bits_per_sample) -{ - switch (bits_per_sample) { - case 8: - return SAMPLE_FORMAT_S8; - - case 16: - return SAMPLE_FORMAT_S16; - - case 24: - return SAMPLE_FORMAT_S24_P32; - - case 32: - return SAMPLE_FORMAT_S32; - - default: - return SAMPLE_FORMAT_UNDEFINED; - } -} - -static void -flac_got_stream_info(struct flac_data *data, - const FLAC__StreamMetadata_StreamInfo *stream_info) -{ - if (data->initialized || data->unsupported) - return; - - GError *error = NULL; - if (!audio_format_init_checked(&data->audio_format, - stream_info->sample_rate, - flac_sample_format(stream_info->bits_per_sample), - stream_info->channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); - data->unsupported = true; - return; - } - - data->frame_size = audio_format_frame_size(&data->audio_format); - - if (data->total_frames == 0) - data->total_frames = stream_info->total_samples; - - data->initialized = true; -} - -void flac_metadata_common_cb(const FLAC__StreamMetadata * block, - struct flac_data *data) -{ - if (data->unsupported) - return; - - struct replay_gain_info rgi; - char *mixramp_start; - char *mixramp_end; - float replay_gain_db = 0; - - switch (block->type) { - case FLAC__METADATA_TYPE_STREAMINFO: - flac_got_stream_info(data, &block->data.stream_info); - break; - - case FLAC__METADATA_TYPE_VORBIS_COMMENT: - if (flac_parse_replay_gain(&rgi, block)) - replay_gain_db = decoder_replay_gain(data->decoder, &rgi); - - if (flac_parse_mixramp(&mixramp_start, &mixramp_end, block)) - decoder_mixramp(data->decoder, replay_gain_db, - mixramp_start, mixramp_end); - - if (data->tag != NULL) - flac_vorbis_comments_to_tag(data->tag, NULL, - &block->data.vorbis_comment); - - default: - break; - } -} - -void flac_error_common_cb(const FLAC__StreamDecoderErrorStatus status, - struct flac_data *data) -{ - if (decoder_get_command(data->decoder) == DECODE_COMMAND_STOP) - return; - - g_warning("%s", FLAC__StreamDecoderErrorStatusString[status]); -} - -/** - * This function attempts to call decoder_initialized() in case there - * was no STREAMINFO block. This is allowed for nonseekable streams, - * where the server sends us only a part of the file, without - * providing the STREAMINFO block from the beginning of the file - * (e.g. when seeking with SqueezeBox Server). - */ -static bool -flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header) -{ - if (data->unsupported) - return false; - - GError *error = NULL; - if (!audio_format_init_checked(&data->audio_format, - header->sample_rate, - flac_sample_format(header->bits_per_sample), - header->channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); - data->unsupported = true; - return false; - } - - data->frame_size = audio_format_frame_size(&data->audio_format); - - decoder_initialized(data->decoder, &data->audio_format, - data->input_stream->seekable, - (float)data->total_frames / - (float)data->audio_format.sample_rate); - - data->initialized = true; - - return true; -} - -FLAC__StreamDecoderWriteStatus -flac_common_write(struct flac_data *data, const FLAC__Frame * frame, - const FLAC__int32 *const buf[], - FLAC__uint64 nbytes) -{ - enum decoder_command cmd; - void *buffer; - unsigned bit_rate; - - if (!data->initialized && !flac_got_first_frame(data, &frame->header)) - return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - - size_t buffer_size = frame->header.blocksize * data->frame_size; - buffer = pcm_buffer_get(&data->buffer, buffer_size); - - flac_convert(buffer, frame->header.channels, - data->audio_format.format, buf, - 0, frame->header.blocksize); - - if (nbytes > 0) - bit_rate = nbytes * 8 * frame->header.sample_rate / - (1000 * frame->header.blocksize); - else - bit_rate = 0; - - cmd = decoder_data(data->decoder, data->input_stream, - buffer, buffer_size, - bit_rate); - data->next_frame += frame->header.blocksize; - switch (cmd) { - case DECODE_COMMAND_NONE: - case DECODE_COMMAND_START: - break; - - case DECODE_COMMAND_STOP: - return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - - case DECODE_COMMAND_SEEK: - return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; - } - - return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; -} diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h deleted file mode 100644 index 0d90ba656..000000000 --- a/src/decoder/_flac_common.h +++ /dev/null @@ -1,105 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* - * Common data structures and functions used by FLAC and OggFLAC - */ - -#ifndef MPD_FLAC_COMMON_H -#define MPD_FLAC_COMMON_H - -#include "decoder_api.h" -#include "pcm_buffer.h" - -#include <glib.h> - -#include <FLAC/stream_decoder.h> -#include <FLAC/metadata.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "flac" - -struct flac_data { - struct pcm_buffer buffer; - - /** - * The size of one frame in the output buffer. - */ - unsigned frame_size; - - /** - * Has decoder_initialized() been called yet? - */ - bool initialized; - - /** - * Does the FLAC file contain an unsupported audio format? - */ - bool unsupported; - - /** - * The validated audio format of the FLAC file. This - * attribute is defined if "initialized" is true. - */ - struct audio_format audio_format; - - /** - * The total number of frames in this song. The decoder - * plugin may initialize this attribute to override the value - * provided by libFLAC (e.g. for sub songs from a CUE sheet). - */ - FLAC__uint64 total_frames; - - /** - * The number of the first frame in this song. This is only - * non-zero if playing sub songs from a CUE sheet. - */ - FLAC__uint64 first_frame; - - /** - * The number of the next frame which is going to be decoded. - */ - FLAC__uint64 next_frame; - - FLAC__uint64 position; - struct decoder *decoder; - struct input_stream *input_stream; - struct tag *tag; -}; - -/* initializes a given FlacData struct */ -void -flac_data_init(struct flac_data *data, struct decoder * decoder, - struct input_stream *input_stream); - -void -flac_data_deinit(struct flac_data *data); - -void flac_metadata_common_cb(const FLAC__StreamMetadata * block, - struct flac_data *data); - -void flac_error_common_cb(FLAC__StreamDecoderErrorStatus status, - struct flac_data *data); - -FLAC__StreamDecoderWriteStatus -flac_common_write(struct flac_data *data, const FLAC__Frame * frame, - const FLAC__int32 *const buf[], - FLAC__uint64 nbytes); - -#endif /* _FLAC_COMMON_H */ diff --git a/src/decoder/_ogg_common.c b/src/decoder/_ogg_common.c deleted file mode 100644 index 09d2712da..000000000 --- a/src/decoder/_ogg_common.c +++ /dev/null @@ -1,46 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* - * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC) - */ - -#include "config.h" -#include "_ogg_common.h" - -ogg_stream_type ogg_stream_type_detect(struct input_stream *inStream) -{ - /* oggflac detection based on code in ogg123 and this post - * http://lists.xiph.org/pipermail/flac/2004-December/000393.html - * ogg123 trunk still doesn't have this patch as of June 2005 */ - unsigned char buf[41]; - size_t r; - - r = decoder_read(NULL, inStream, buf, sizeof(buf)); - if (r < sizeof(buf) || memcmp(buf, "OggS", 4) != 0) - return VORBIS; - - if ((memcmp(buf + 29, "FLAC", 4) == 0 && - memcmp(buf + 37, "fLaC", 4) == 0) || - memcmp(buf + 28, "FLAC", 4) == 0 || - memcmp(buf + 28, "fLaC", 4) == 0) - return FLAC; - - return VORBIS; -} diff --git a/src/decoder/_ogg_common.h b/src/decoder/_ogg_common.h deleted file mode 100644 index 85e4ebba6..000000000 --- a/src/decoder/_ogg_common.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* - * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC) - */ - -#ifndef MPD_OGG_COMMON_H -#define MPD_OGG_COMMON_H - -#include "decoder_api.h" - -typedef enum _ogg_stream_type { VORBIS, FLAC } ogg_stream_type; - -ogg_stream_type ogg_stream_type_detect(struct input_stream *inStream); - -#endif /* _OGG_COMMON_H */ diff --git a/src/decoder/audiofile_decoder_plugin.c b/src/decoder/audiofile_decoder_plugin.c deleted file mode 100644 index b344795e7..000000000 --- a/src/decoder/audiofile_decoder_plugin.c +++ /dev/null @@ -1,258 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "tag_handler.h" - -#include <audiofile.h> -#include <af_vfs.h> -#include <assert.h> -#include <glib.h> -#include <stdio.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "audiofile" - -/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */ -#define CHUNK_SIZE 1020 - -static int audiofile_get_duration(const char *file) -{ - int total_time; - AFfilehandle af_fp = afOpenFile(file, "r", NULL); - if (af_fp == AF_NULL_FILEHANDLE) { - return -1; - } - total_time = (int) - ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK) - / afGetRate(af_fp, AF_DEFAULT_TRACK)); - afCloseFile(af_fp); - return total_time; -} - -static ssize_t -audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length) -{ - struct input_stream *is = (struct input_stream *) vfile->closure; - GError *error = NULL; - size_t nbytes; - - nbytes = input_stream_lock_read(is, data, length, &error); - if (nbytes == 0 && error != NULL) { - g_warning("%s", error->message); - g_error_free(error); - return -1; - } - - return nbytes; -} - -static AFfileoffset -audiofile_file_length(AFvirtualfile *vfile) -{ - struct input_stream *is = (struct input_stream *) vfile->closure; - return is->size; -} - -static AFfileoffset -audiofile_file_tell(AFvirtualfile *vfile) -{ - struct input_stream *is = (struct input_stream *) vfile->closure; - return is->offset; -} - -static void -audiofile_file_destroy(AFvirtualfile *vfile) -{ - assert(vfile->closure != NULL); - - vfile->closure = NULL; -} - -static AFfileoffset -audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset offset, int is_relative) -{ - struct input_stream *is = (struct input_stream *) vfile->closure; - int whence = (is_relative ? SEEK_CUR : SEEK_SET); - if (input_stream_lock_seek(is, offset, whence, NULL)) { - return is->offset; - } else { - return -1; - } -} - -static AFvirtualfile * -setup_virtual_fops(struct input_stream *stream) -{ - AFvirtualfile *vf = g_malloc(sizeof(AFvirtualfile)); - vf->closure = stream; - vf->write = NULL; - vf->read = audiofile_file_read; - vf->length = audiofile_file_length; - vf->destroy = audiofile_file_destroy; - vf->seek = audiofile_file_seek; - vf->tell = audiofile_file_tell; - return vf; -} - -static enum sample_format -audiofile_bits_to_sample_format(int bits) -{ - switch (bits) { - case 8: - return SAMPLE_FORMAT_S8; - - case 16: - return SAMPLE_FORMAT_S16; - - case 24: - return SAMPLE_FORMAT_S24_P32; - - case 32: - return SAMPLE_FORMAT_S32; - } - - return SAMPLE_FORMAT_UNDEFINED; -} - -static enum sample_format -audiofile_setup_sample_format(AFfilehandle af_fp) -{ - int fs, bits; - - afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - if (!audio_valid_sample_format(audiofile_bits_to_sample_format(bits))) { - g_debug("input file has %d bit samples, converting to 16", - bits); - bits = 16; - } - - afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, - AF_SAMPFMT_TWOSCOMP, bits); - afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - - return audiofile_bits_to_sample_format(bits); -} - -static void -audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) -{ - GError *error = NULL; - AFvirtualfile *vf; - int fs, frame_count; - AFfilehandle af_fp; - struct audio_format audio_format; - float total_time; - uint16_t bit_rate; - int ret; - char chunk[CHUNK_SIZE]; - enum decoder_command cmd; - - if (!is->seekable) { - g_warning("not seekable"); - return; - } - - vf = setup_virtual_fops(is); - - af_fp = afOpenVirtualFile(vf, "r", NULL); - if (af_fp == AF_NULL_FILEHANDLE) { - g_warning("failed to input stream\n"); - return; - } - - if (!audio_format_init_checked(&audio_format, - afGetRate(af_fp, AF_DEFAULT_TRACK), - audiofile_setup_sample_format(af_fp), - afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK), - &error)) { - g_warning("%s", error->message); - g_error_free(error); - afCloseFile(af_fp); - return; - } - - frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); - - total_time = ((float)frame_count / (float)audio_format.sample_rate); - - bit_rate = (uint16_t)(is->size * 8.0 / total_time / 1000.0 + 0.5); - - fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); - - decoder_initialized(decoder, &audio_format, true, total_time); - - do { - ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, - CHUNK_SIZE / fs); - if (ret <= 0) - break; - - cmd = decoder_data(decoder, NULL, - chunk, ret * fs, - bit_rate); - - if (cmd == DECODE_COMMAND_SEEK) { - AFframecount frame = decoder_seek_where(decoder) * - audio_format.sample_rate; - afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame); - - decoder_command_finished(decoder); - cmd = DECODE_COMMAND_NONE; - } - } while (cmd == DECODE_COMMAND_NONE); - - afCloseFile(af_fp); -} - -static bool -audiofile_scan_file(const char *file, - const struct tag_handler *handler, void *handler_ctx) -{ - int total_time = audiofile_get_duration(file); - - if (total_time < 0) { - g_debug("Failed to get total song time from: %s\n", - file); - return false; - } - - tag_handler_invoke_duration(handler, handler_ctx, total_time); - return true; -} - -static const char *const audiofile_suffixes[] = { - "wav", "au", "aiff", "aif", NULL -}; - -static const char *const audiofile_mime_types[] = { - "audio/x-wav", - "audio/x-aiff", - NULL -}; - -const struct decoder_plugin audiofile_decoder_plugin = { - .name = "audiofile", - .stream_decode = audiofile_stream_decode, - .scan_file = audiofile_scan_file, - .suffixes = audiofile_suffixes, - .mime_types = audiofile_mime_types, -}; diff --git a/src/decoder/dsdiff_decoder_plugin.c b/src/decoder/dsdiff_decoder_plugin.c deleted file mode 100644 index 84471fb3a..000000000 --- a/src/decoder/dsdiff_decoder_plugin.c +++ /dev/null @@ -1,397 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* \file - * - * This plugin decodes DSDIFF data (SACD) embedded in DFF files. - * The DFF code was modeled after the specification found here: - * http://www.sonicstudio.com/pdf/dsd/DSDIFF_1.5_Spec.pdf - * - * All functions common to both DSD decoders have been moved to dsdlib - */ - -#include "config.h" -#include "dsdiff_decoder_plugin.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "util/bit_reverse.h" -#include "tag_handler.h" -#include "dsdlib.h" -#include "tag_handler.h" - -#include <unistd.h> -#include <stdio.h> /* for SEEK_SET, SEEK_CUR */ - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "dsdiff" - -struct dsdiff_header { - struct dsdlib_id id; - uint32_t size_high, size_low; - struct dsdlib_id format; -}; - -struct dsdiff_chunk_header { - struct dsdlib_id id; - uint32_t size_high, size_low; -}; - -struct dsdiff_metadata { - unsigned sample_rate, channels; - bool bitreverse; - uint64_t chunk_size; -}; - -static bool lsbitfirst; - -static bool -dsdiff_init(const struct config_param *param) -{ - lsbitfirst = config_get_block_bool(param, "lsbitfirst", false); - return true; -} - -/** - * Read the "size" attribute from the specified header, converting it - * to the host byte order if needed. - */ -G_GNUC_CONST -static uint64_t -dsdiff_chunk_size(const struct dsdiff_chunk_header *header) -{ - return (((uint64_t)GUINT32_FROM_BE(header->size_high)) << 32) | - ((uint64_t)GUINT32_FROM_BE(header->size_low)); -} - -static bool -dsdiff_read_id(struct decoder *decoder, struct input_stream *is, - struct dsdlib_id *id) -{ - return dsdlib_read(decoder, is, id, sizeof(*id)); -} - -static bool -dsdiff_read_chunk_header(struct decoder *decoder, struct input_stream *is, - struct dsdiff_chunk_header *header) -{ - return dsdlib_read(decoder, is, header, sizeof(*header)); -} - -static bool -dsdiff_read_payload(struct decoder *decoder, struct input_stream *is, - const struct dsdiff_chunk_header *header, - void *data, size_t length) -{ - uint64_t size = dsdiff_chunk_size(header); - if (size != (uint64_t)length) - return false; - - size_t nbytes = decoder_read(decoder, is, data, length); - return nbytes == length; -} - -/** - * Read and parse a "SND" chunk inside "PROP". - */ -static bool -dsdiff_read_prop_snd(struct decoder *decoder, struct input_stream *is, - struct dsdiff_metadata *metadata, - goffset end_offset) -{ - struct dsdiff_chunk_header header; - while ((goffset)(is->offset + sizeof(header)) <= end_offset) { - if (!dsdiff_read_chunk_header(decoder, is, &header)) - return false; - - goffset chunk_end_offset = - is->offset + dsdiff_chunk_size(&header); - if (chunk_end_offset > end_offset) - return false; - - if (dsdlib_id_equals(&header.id, "FS ")) { - uint32_t sample_rate; - if (!dsdiff_read_payload(decoder, is, &header, - &sample_rate, - sizeof(sample_rate))) - return false; - - metadata->sample_rate = GUINT32_FROM_BE(sample_rate); - } else if (dsdlib_id_equals(&header.id, "CHNL")) { - uint16_t channels; - if (dsdiff_chunk_size(&header) < sizeof(channels) || - !dsdlib_read(decoder, is, - &channels, sizeof(channels)) || - !dsdlib_skip_to(decoder, is, chunk_end_offset)) - return false; - - metadata->channels = GUINT16_FROM_BE(channels); - } else if (dsdlib_id_equals(&header.id, "CMPR")) { - struct dsdlib_id type; - if (dsdiff_chunk_size(&header) < sizeof(type) || - !dsdlib_read(decoder, is, - &type, sizeof(type)) || - !dsdlib_skip_to(decoder, is, chunk_end_offset)) - return false; - - if (!dsdlib_id_equals(&type, "DSD ")) - /* only uncompressed DSD audio data - is implemented */ - return false; - } else { - /* ignore unknown chunk */ - - if (!dsdlib_skip_to(decoder, is, chunk_end_offset)) - return false; - } - } - - return is->offset == end_offset; -} - -/** - * Read and parse a "PROP" chunk. - */ -static bool -dsdiff_read_prop(struct decoder *decoder, struct input_stream *is, - struct dsdiff_metadata *metadata, - const struct dsdiff_chunk_header *prop_header) -{ - uint64_t prop_size = dsdiff_chunk_size(prop_header); - goffset end_offset = is->offset + prop_size; - - struct dsdlib_id prop_id; - if (prop_size < sizeof(prop_id) || - !dsdiff_read_id(decoder, is, &prop_id)) - return false; - - if (dsdlib_id_equals(&prop_id, "SND ")) - return dsdiff_read_prop_snd(decoder, is, metadata, end_offset); - else - /* ignore unknown PROP chunk */ - return dsdlib_skip_to(decoder, is, end_offset); -} - -/** - * Read and parse all metadata chunks at the beginning. Stop when the - * first "DSD" chunk is seen, and return its header in the - * "chunk_header" parameter. - */ -static bool -dsdiff_read_metadata(struct decoder *decoder, struct input_stream *is, - struct dsdiff_metadata *metadata, - struct dsdiff_chunk_header *chunk_header) -{ - struct dsdiff_header header; - if (!dsdlib_read(decoder, is, &header, sizeof(header)) || - !dsdlib_id_equals(&header.id, "FRM8") || - !dsdlib_id_equals(&header.format, "DSD ")) - return false; - - while (true) { - if (!dsdiff_read_chunk_header(decoder, is, - chunk_header)) - return false; - - if (dsdlib_id_equals(&chunk_header->id, "PROP")) { - if (!dsdiff_read_prop(decoder, is, metadata, - chunk_header)) - return false; - } else if (dsdlib_id_equals(&chunk_header->id, "DSD ")) { - uint64_t chunk_size; - chunk_size = dsdiff_chunk_size(chunk_header); - metadata->chunk_size = chunk_size; - return true; - } else { - /* ignore unknown chunk */ - uint64_t chunk_size; - chunk_size = dsdiff_chunk_size(chunk_header); - goffset chunk_end_offset = is->offset + chunk_size; - - if (!dsdlib_skip_to(decoder, is, chunk_end_offset)) - return false; - } - } -} - -static void -bit_reverse_buffer(uint8_t *p, uint8_t *end) -{ - for (; p < end; ++p) - *p = bit_reverse(*p); -} - -/** - * Decode one "DSD" chunk. - */ -static bool -dsdiff_decode_chunk(struct decoder *decoder, struct input_stream *is, - unsigned channels, - uint64_t chunk_size) -{ - uint8_t buffer[8192]; - - const size_t sample_size = sizeof(buffer[0]); - const size_t frame_size = channels * sample_size; - const unsigned buffer_frames = sizeof(buffer) / frame_size; - const unsigned buffer_samples = buffer_frames * frame_size; - const size_t buffer_size = buffer_samples * sample_size; - - while (chunk_size > 0) { - /* see how much aligned data from the remaining chunk - fits into the local buffer */ - unsigned now_frames = buffer_frames; - size_t now_size = buffer_size; - if (chunk_size < (uint64_t)now_size) { - now_frames = (unsigned)chunk_size / frame_size; - now_size = now_frames * frame_size; - } - - size_t nbytes = decoder_read(decoder, is, buffer, now_size); - if (nbytes != now_size) - return false; - - chunk_size -= nbytes; - - if (lsbitfirst) - bit_reverse_buffer(buffer, buffer + nbytes); - - enum decoder_command cmd = - decoder_data(decoder, is, buffer, nbytes, 0); - switch (cmd) { - case DECODE_COMMAND_NONE: - break; - - case DECODE_COMMAND_START: - case DECODE_COMMAND_STOP: - return false; - - case DECODE_COMMAND_SEEK: - - /* Not implemented yet */ - decoder_seek_error(decoder); - break; - } - } - return dsdlib_skip(decoder, is, chunk_size); -} - -static void -dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is) -{ - struct dsdiff_metadata metadata = { - .sample_rate = 0, - .channels = 0, - }; - - struct dsdiff_chunk_header chunk_header; - /* check if it is is a proper DFF file */ - if (!dsdiff_read_metadata(decoder, is, &metadata, &chunk_header)) - return; - - GError *error = NULL; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, - metadata.channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); - return; - } - - /* calculate song time from DSD chunk size and sample frequency */ - uint64_t chunk_size = metadata.chunk_size; - float songtime = ((chunk_size / metadata.channels) * 8) / - (float) metadata.sample_rate; - - /* success: file was recognized */ - decoder_initialized(decoder, &audio_format, false, songtime); - - /* every iteration of the following loop decodes one "DSD" - chunk from a DFF file */ - - while (true) { - chunk_size = dsdiff_chunk_size(&chunk_header); - - if (dsdlib_id_equals(&chunk_header.id, "DSD ")) { - if (!dsdiff_decode_chunk(decoder, is, - metadata.channels, - chunk_size)) - break; - } else { - /* ignore other chunks */ - if (!dsdlib_skip(decoder, is, chunk_size)) - break; - } - - /* read next chunk header; the first one was read by - dsdiff_read_metadata() */ - if (!dsdiff_read_chunk_header(decoder, - is, &chunk_header)) - break; - } -} - -static bool -dsdiff_scan_stream(struct input_stream *is, - G_GNUC_UNUSED const struct tag_handler *handler, - G_GNUC_UNUSED void *handler_ctx) -{ - struct dsdiff_metadata metadata = { - .sample_rate = 0, - .channels = 0, - }; - - struct dsdiff_chunk_header chunk_header; - /* First check for DFF metadata */ - if (!dsdiff_read_metadata(NULL, is, &metadata, &chunk_header)) - return false; - - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, - metadata.channels, NULL)) - /* refuse to parse files which we cannot play anyway */ - return false; - - /* calculate song time and add as tag */ - unsigned songtime = ((metadata.chunk_size / metadata.channels) * 8) / - metadata.sample_rate; - tag_handler_invoke_duration(handler, handler_ctx, songtime); - - return true; -} - -static const char *const dsdiff_suffixes[] = { - "dff", - NULL -}; - -static const char *const dsdiff_mime_types[] = { - "application/x-dff", - NULL -}; - -const struct decoder_plugin dsdiff_decoder_plugin = { - .name = "dsdiff", - .init = dsdiff_init, - .stream_decode = dsdiff_stream_decode, - .scan_stream = dsdiff_scan_stream, - .suffixes = dsdiff_suffixes, - .mime_types = dsdiff_mime_types, -}; diff --git a/src/decoder/dsdiff_decoder_plugin.h b/src/decoder/dsdiff_decoder_plugin.h deleted file mode 100644 index 452f9050b..000000000 --- a/src/decoder/dsdiff_decoder_plugin.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_DSDIFF_H -#define MPD_DECODER_DSDIFF_H - -extern const struct decoder_plugin dsdiff_decoder_plugin; - -#endif diff --git a/src/decoder/dsdlib.c b/src/decoder/dsdlib.c deleted file mode 100644 index 3df9497c4..000000000 --- a/src/decoder/dsdlib.c +++ /dev/null @@ -1,112 +0,0 @@ -/* - * Copyright (C) 2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* \file - * - * This file contains functions used by the DSF and DSDIFF decoders. - * - */ - -#include "config.h" -#include "dsf_decoder_plugin.h" -#include "decoder_api.h" -#include "util/bit_reverse.h" -#include "dsdlib.h" -#include "dsdiff_decoder_plugin.h" - -#include <unistd.h> -#include <stdio.h> /* for SEEK_SET, SEEK_CUR */ - -bool -dsdlib_id_equals(const struct dsdlib_id *id, const char *s) -{ - assert(id != NULL); - assert(s != NULL); - assert(strlen(s) == sizeof(id->value)); - - return memcmp(id->value, s, sizeof(id->value)) == 0; -} - -bool -dsdlib_read(struct decoder *decoder, struct input_stream *is, - void *data, size_t length) -{ - size_t nbytes = decoder_read(decoder, is, data, length); - return nbytes == length; -} - -/** - * Skip the #input_stream to the specified offset. - */ -bool -dsdlib_skip_to(struct decoder *decoder, struct input_stream *is, - goffset offset) -{ - if (is->seekable) - return input_stream_seek(is, offset, SEEK_SET, NULL); - - if (is->offset > offset) - return false; - - char buffer[8192]; - while (is->offset < offset) { - size_t length = sizeof(buffer); - if (offset - is->offset < (goffset)length) - length = offset - is->offset; - - size_t nbytes = decoder_read(decoder, is, buffer, length); - if (nbytes == 0) - return false; - } - - assert(is->offset == offset); - return true; -} - -/** - * Skip some bytes from the #input_stream. - */ -bool -dsdlib_skip(struct decoder *decoder, struct input_stream *is, - goffset delta) -{ - assert(delta >= 0); - - if (delta == 0) - return true; - - if (is->seekable) - return input_stream_seek(is, delta, SEEK_CUR, NULL); - - char buffer[8192]; - while (delta > 0) { - size_t length = sizeof(buffer); - if ((goffset)length > delta) - length = delta; - - size_t nbytes = decoder_read(decoder, is, buffer, length); - if (nbytes == 0) - return false; - - delta -= nbytes; - } - - return true; -} - diff --git a/src/decoder/dsdlib.h b/src/decoder/dsdlib.h deleted file mode 100644 index d9675f5fe..000000000 --- a/src/decoder/dsdlib.h +++ /dev/null @@ -1,42 +0,0 @@ -/* - * Copyright (C) 2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_DSDLIB_H -#define MPD_DECODER_DSDLIB_H - -struct dsdlib_id { - char value[4]; -}; - -bool -dsdlib_id_equals(const struct dsdlib_id *id, const char *s); - -bool -dsdlib_read(struct decoder *decoder, struct input_stream *is, - void *data, size_t length); - -bool -dsdlib_skip_to(struct decoder *decoder, struct input_stream *is, - goffset offset); - -bool -dsdlib_skip(struct decoder *decoder, struct input_stream *is, - goffset delta); - -#endif diff --git a/src/decoder/dsf_decoder_plugin.c b/src/decoder/dsf_decoder_plugin.c deleted file mode 100644 index c0107eb30..000000000 --- a/src/decoder/dsf_decoder_plugin.c +++ /dev/null @@ -1,338 +0,0 @@ -/* - * Copyright (C) 2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* \file - * - * This plugin decodes DSDIFF data (SACD) embedded in DSF files. - * - * The DSF code was created using the specification found here: - * http://dsd-guide.com/sonys-dsf-file-format-spec - * - * All functions common to both DSD decoders have been moved to dsdlib - */ - -#include "config.h" -#include "dsf_decoder_plugin.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "util/bit_reverse.h" -#include "dsdlib.h" -#include "tag_handler.h" - -#include <unistd.h> -#include <stdio.h> /* for SEEK_SET, SEEK_CUR */ - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "dsf" - -struct dsf_metadata { - unsigned sample_rate, channels; - bool bitreverse; - uint64_t chunk_size; -}; - -struct dsf_header { - /** DSF header id: "DSD " */ - struct dsdlib_id id; - /** DSD chunk size, including id = 28 */ - uint32_t size_low, size_high; - /** total file size */ - uint32_t fsize_low, fsize_high; - /** pointer to id3v2 metadata, should be at the end of the file */ - uint32_t pmeta_low, pmeta_high; -}; -/** DSF file fmt chunk */ -struct dsf_fmt_chunk { - - /** id: "fmt " */ - struct dsdlib_id id; - /** fmt chunk size, including id, normally 52 */ - uint32_t size_low, size_high; - /** version of this format = 1 */ - uint32_t version; - /** 0: DSD raw */ - uint32_t formatid; - /** channel type, 1 = mono, 2 = stereo, 3 = 3 channels, etc */ - uint32_t channeltype; - /** Channel number, 1 = mono, 2 = stereo, ... 6 = 6 channels */ - uint32_t channelnum; - /** sample frequency: 2822400, 5644800 */ - uint32_t sample_freq; - /** bits per sample 1 or 8 */ - uint32_t bitssample; - /** Sample count per channel in bytes */ - uint32_t scnt_low, scnt_high; - /** block size per channel = 4096 */ - uint32_t block_size; - /** reserved, should be all zero */ - uint32_t reserved; -}; - -struct dsf_data_chunk { - struct dsdlib_id id; - /** "data" chunk size, includes header (id+size) */ - uint32_t size_low, size_high; -}; - -/** - * Read and parse all needed metadata chunks for DSF files. - */ -static bool -dsf_read_metadata(struct decoder *decoder, struct input_stream *is, - struct dsf_metadata *metadata) -{ - uint64_t chunk_size; - struct dsf_header dsf_header; - if (!dsdlib_read(decoder, is, &dsf_header, sizeof(dsf_header)) || - !dsdlib_id_equals(&dsf_header.id, "DSD ")) - return false; - - chunk_size = (((uint64_t)GUINT32_FROM_LE(dsf_header.size_high)) << 32) | - ((uint64_t)GUINT32_FROM_LE(dsf_header.size_low)); - - if (sizeof(dsf_header) != chunk_size) - return false; - - /* read the 'fmt ' chunk of the DSF file */ - struct dsf_fmt_chunk dsf_fmt_chunk; - if (!dsdlib_read(decoder, is, &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) || - !dsdlib_id_equals(&dsf_fmt_chunk.id, "fmt ")) - return false; - - uint64_t fmt_chunk_size; - fmt_chunk_size = (((uint64_t)GUINT32_FROM_LE(dsf_fmt_chunk.size_high)) << 32) | - ((uint64_t)GUINT32_FROM_LE(dsf_fmt_chunk.size_low)); - - if (fmt_chunk_size != sizeof(dsf_fmt_chunk)) - return false; - - uint32_t samplefreq = (uint32_t)GUINT32_FROM_LE(dsf_fmt_chunk.sample_freq); - - /* for now, only support version 1 of the standard, DSD raw stereo - files with a sample freq of 2822400 Hz */ - - if (dsf_fmt_chunk.version != 1 || dsf_fmt_chunk.formatid != 0 - || dsf_fmt_chunk.channeltype != 2 - || dsf_fmt_chunk.channelnum != 2 - || samplefreq != 2822400) - return false; - - uint32_t chblksize = (uint32_t)GUINT32_FROM_LE(dsf_fmt_chunk.block_size); - /* according to the spec block size should always be 4096 */ - if (chblksize != 4096) - return false; - - /* read the 'data' chunk of the DSF file */ - struct dsf_data_chunk data_chunk; - if (!dsdlib_read(decoder, is, &data_chunk, sizeof(data_chunk)) || - !dsdlib_id_equals(&data_chunk.id, "data")) - return false; - - /* data size of DSF files are padded to multiple of 4096, - we use the actual data size as chunk size */ - - uint64_t data_size; - data_size = (((uint64_t)GUINT32_FROM_LE(data_chunk.size_high)) << 32) | - ((uint64_t)GUINT32_FROM_LE(data_chunk.size_low)); - data_size -= sizeof(data_chunk); - - metadata->chunk_size = data_size; - metadata->channels = (unsigned) dsf_fmt_chunk.channelnum; - metadata->sample_rate = samplefreq; - - /* check bits per sample format, determine if bitreverse is needed */ - metadata->bitreverse = dsf_fmt_chunk.bitssample == 1; - return true; -} - -static void -bit_reverse_buffer(uint8_t *p, uint8_t *end) -{ - for (; p < end; ++p) - *p = bit_reverse(*p); -} - -/** - * DSF data is build up of alternating 4096 blocks of DSD samples for left and - * right. Convert the buffer holding 1 block of 4096 DSD left samples and 1 - * block of 4096 DSD right samples to 8k of samples in normal PCM left/right - * order. - */ -static void -dsf_to_pcm_order(uint8_t *dest, uint8_t *scratch, size_t nrbytes) -{ - for (unsigned i = 0, j = 0; i < (unsigned)nrbytes; i += 2) { - scratch[i] = *(dest+j); - j++; - } - - for (unsigned i = 1, j = 0; i < (unsigned) nrbytes; i += 2) { - scratch[i] = *(dest+4096+j); - j++; - } - - for (unsigned i = 0; i < (unsigned)nrbytes; i++) { - *dest = scratch[i]; - dest++; - } -} - -/** - * Decode one complete DSF 'data' chunk i.e. a complete song - */ -static bool -dsf_decode_chunk(struct decoder *decoder, struct input_stream *is, - unsigned channels, - uint64_t chunk_size, - bool bitreverse) -{ - uint8_t buffer[8192]; - - /* scratch buffer for DSF samples to convert to the needed - normal left/right regime of samples */ - uint8_t dsf_scratch_buffer[8192]; - - const size_t sample_size = sizeof(buffer[0]); - const size_t frame_size = channels * sample_size; - const unsigned buffer_frames = sizeof(buffer) / frame_size; - const unsigned buffer_samples = buffer_frames * frame_size; - const size_t buffer_size = buffer_samples * sample_size; - - while (chunk_size > 0) { - /* see how much aligned data from the remaining chunk - fits into the local buffer */ - unsigned now_frames = buffer_frames; - size_t now_size = buffer_size; - if (chunk_size < (uint64_t)now_size) { - now_frames = (unsigned)chunk_size / frame_size; - now_size = now_frames * frame_size; - } - - size_t nbytes = decoder_read(decoder, is, buffer, now_size); - if (nbytes != now_size) - return false; - - chunk_size -= nbytes; - - if (bitreverse) - bit_reverse_buffer(buffer, buffer + nbytes); - - dsf_to_pcm_order(buffer, dsf_scratch_buffer, nbytes); - - enum decoder_command cmd = - decoder_data(decoder, is, buffer, nbytes, 0); - switch (cmd) { - case DECODE_COMMAND_NONE: - break; - - case DECODE_COMMAND_START: - case DECODE_COMMAND_STOP: - return false; - - case DECODE_COMMAND_SEEK: - - /* not implemented yet */ - decoder_seek_error(decoder); - break; - } - } - return dsdlib_skip(decoder, is, chunk_size); -} - -static void -dsf_stream_decode(struct decoder *decoder, struct input_stream *is) -{ - struct dsf_metadata metadata = { - .sample_rate = 0, - .channels = 0, - }; - - /* check if it is a proper DSF file */ - if (!dsf_read_metadata(decoder, is, &metadata)) - return; - - GError *error = NULL; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, - metadata.channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); - return; - } - /* Calculate song time from DSD chunk size and sample frequency */ - uint64_t chunk_size = metadata.chunk_size; - float songtime = ((chunk_size / metadata.channels) * 8) / - (float) metadata.sample_rate; - - /* success: file was recognized */ - decoder_initialized(decoder, &audio_format, false, songtime); - - if (!dsf_decode_chunk(decoder, is, metadata.channels, - metadata.chunk_size, - metadata.bitreverse)) - return; -} - -static bool -dsf_scan_stream(struct input_stream *is, - G_GNUC_UNUSED const struct tag_handler *handler, - G_GNUC_UNUSED void *handler_ctx) -{ - struct dsf_metadata metadata = { - .sample_rate = 0, - .channels = 0, - }; - - /* check DSF metadata */ - if (!dsf_read_metadata(NULL, is, &metadata)) - return false; - - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8, - SAMPLE_FORMAT_DSD, - metadata.channels, NULL)) - /* refuse to parse files which we cannot play anyway */ - return false; - - /* calculate song time and add as tag */ - unsigned songtime = ((metadata.chunk_size / metadata.channels) * 8) / - metadata.sample_rate; - tag_handler_invoke_duration(handler, handler_ctx, songtime); - - return true; -} - -static const char *const dsf_suffixes[] = { - "dsf", - NULL -}; - -static const char *const dsf_mime_types[] = { - "application/x-dsf", - NULL -}; - -const struct decoder_plugin dsf_decoder_plugin = { - .name = "dsf", - .stream_decode = dsf_stream_decode, - .scan_stream = dsf_scan_stream, - .suffixes = dsf_suffixes, - .mime_types = dsf_mime_types, -}; diff --git a/src/decoder/dsf_decoder_plugin.h b/src/decoder/dsf_decoder_plugin.h deleted file mode 100644 index 401d3fed7..000000000 --- a/src/decoder/dsf_decoder_plugin.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * Copyright (C) 2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_DSF_H -#define MPD_DECODER_DSF_H - -extern const struct decoder_plugin dsf_decoder_plugin; - -#endif diff --git a/src/decoder/faad_decoder_plugin.c b/src/decoder/faad_decoder_plugin.c deleted file mode 100644 index 911f033b8..000000000 --- a/src/decoder/faad_decoder_plugin.c +++ /dev/null @@ -1,515 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "decoder_buffer.h" -#include "audio_check.h" -#include "tag_handler.h" - -#define AAC_MAX_CHANNELS 6 - -#include <assert.h> -#include <unistd.h> -#include <faad.h> -#include <glib.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "faad" - -static const unsigned adts_sample_rates[] = - { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, - 16000, 12000, 11025, 8000, 7350, 0, 0, 0 -}; - -/** - * The GLib quark used for errors reported by this plugin. - */ -static inline GQuark -faad_decoder_quark(void) -{ - return g_quark_from_static_string("faad"); -} - -/** - * Check whether the buffer head is an AAC frame, and return the frame - * length. Returns 0 if it is not a frame. - */ -static size_t -adts_check_frame(const unsigned char *data) -{ - /* check syncword */ - if (!((data[0] == 0xFF) && ((data[1] & 0xF6) == 0xF0))) - return 0; - - return (((unsigned int)data[3] & 0x3) << 11) | - (((unsigned int)data[4]) << 3) | - (data[5] >> 5); -} - -/** - * Find the next AAC frame in the buffer. Returns 0 if no frame is - * found or if not enough data is available. - */ -static size_t -adts_find_frame(struct decoder_buffer *buffer) -{ - const unsigned char *data, *p; - size_t length, frame_length; - bool ret; - - while (true) { - data = decoder_buffer_read(buffer, &length); - if (data == NULL || length < 8) { - /* not enough data yet */ - ret = decoder_buffer_fill(buffer); - if (!ret) - /* failed */ - return 0; - - continue; - } - - /* find the 0xff marker */ - p = memchr(data, 0xff, length); - if (p == NULL) { - /* no marker - discard the buffer */ - decoder_buffer_consume(buffer, length); - continue; - } - - if (p > data) { - /* discard data before 0xff */ - decoder_buffer_consume(buffer, p - data); - continue; - } - - /* is it a frame? */ - frame_length = adts_check_frame(data); - if (frame_length == 0) { - /* it's just some random 0xff byte; discard it - and continue searching */ - decoder_buffer_consume(buffer, 1); - continue; - } - - if (length < frame_length) { - /* available buffer size is smaller than the - frame will be - attempt to read more - data */ - ret = decoder_buffer_fill(buffer); - if (!ret) { - /* not enough data; discard this frame - to prevent a possible buffer - overflow */ - data = decoder_buffer_read(buffer, &length); - if (data != NULL) - decoder_buffer_consume(buffer, length); - } - - continue; - } - - /* found a full frame! */ - return frame_length; - } -} - -static float -adts_song_duration(struct decoder_buffer *buffer) -{ - unsigned int frames, frame_length; - unsigned sample_rate = 0; - float frames_per_second; - - /* Read all frames to ensure correct time and bitrate */ - for (frames = 0;; frames++) { - frame_length = adts_find_frame(buffer); - if (frame_length == 0) - break; - - - if (frames == 0) { - const unsigned char *data; - size_t buffer_length; - - data = decoder_buffer_read(buffer, &buffer_length); - assert(data != NULL); - assert(frame_length <= buffer_length); - - sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2]; - } - - decoder_buffer_consume(buffer, frame_length); - } - - frames_per_second = (float)sample_rate / 1024.0; - if (frames_per_second <= 0) - return -1; - - return (float)frames / frames_per_second; -} - -static float -faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is) -{ - size_t fileread; - size_t tagsize; - const unsigned char *data; - size_t length; - bool success; - - fileread = is->size >= 0 ? is->size : 0; - - decoder_buffer_fill(buffer); - data = decoder_buffer_read(buffer, &length); - if (data == NULL) - return -1; - - tagsize = 0; - if (length >= 10 && !memcmp(data, "ID3", 3)) { - /* skip the ID3 tag */ - - tagsize = (data[6] << 21) | (data[7] << 14) | - (data[8] << 7) | (data[9] << 0); - - tagsize += 10; - - success = decoder_buffer_skip(buffer, tagsize) && - decoder_buffer_fill(buffer); - if (!success) - return -1; - - data = decoder_buffer_read(buffer, &length); - if (data == NULL) - return -1; - } - - if (is->seekable && length >= 2 && - data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) { - /* obtain the duration from the ADTS header */ - float song_length = adts_song_duration(buffer); - - input_stream_lock_seek(is, tagsize, SEEK_SET, NULL); - - data = decoder_buffer_read(buffer, &length); - if (data != NULL) - decoder_buffer_consume(buffer, length); - decoder_buffer_fill(buffer); - - return song_length; - } else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) { - /* obtain the duration from the ADIF header */ - unsigned bit_rate; - size_t skip_size = (data[4] & 0x80) ? 9 : 0; - - if (8 + skip_size > length) - /* not enough data yet; skip parsing this - header */ - return -1; - - bit_rate = ((data[4 + skip_size] & 0x0F) << 19) | - (data[5 + skip_size] << 11) | - (data[6 + skip_size] << 3) | - (data[7 + skip_size] & 0xE0); - - if (fileread != 0 && bit_rate != 0) - return fileread * 8.0 / bit_rate; - else - return fileread; - } else - return -1; -} - -/** - * Wrapper for faacDecInit() which works around some API - * inconsistencies in libfaad. - */ -static bool -faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer, - struct audio_format *audio_format, GError **error_r) -{ - union { - /* deconst hack for libfaad */ - const void *in; - void *out; - } u; - size_t length; - int32_t nbytes; - uint32_t sample_rate; - uint8_t channels; -#ifdef HAVE_FAAD_LONG - /* neaacdec.h declares all arguments as "unsigned long", but - internally expects uint32_t pointers. To avoid gcc - warnings, use this workaround. */ - unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate; -#else - uint32_t *sample_rate_p = &sample_rate; -#endif - - u.in = decoder_buffer_read(buffer, &length); - if (u.in == NULL) { - g_set_error(error_r, faad_decoder_quark(), 0, - "Empty file"); - return false; - } - - nbytes = faacDecInit(decoder, u.out, -#ifdef HAVE_FAAD_BUFLEN_FUNCS - length, -#endif - sample_rate_p, &channels); - if (nbytes < 0) { - g_set_error(error_r, faad_decoder_quark(), 0, - "Not an AAC stream"); - return false; - } - - decoder_buffer_consume(buffer, nbytes); - - return audio_format_init_checked(audio_format, sample_rate, - SAMPLE_FORMAT_S16, channels, error_r); -} - -/** - * Wrapper for faacDecDecode() which works around some API - * inconsistencies in libfaad. - */ -static const void * -faad_decoder_decode(faacDecHandle decoder, struct decoder_buffer *buffer, - faacDecFrameInfo *frame_info) -{ - union { - /* deconst hack for libfaad */ - const void *in; - void *out; - } u; - size_t length; - void *result; - - u.in = decoder_buffer_read(buffer, &length); - if (u.in == NULL) - return NULL; - - result = faacDecDecode(decoder, frame_info, - u.out -#ifdef HAVE_FAAD_BUFLEN_FUNCS - , length -#endif - ); - - return result; -} - -/** - * Get a song file's total playing time in seconds, as a float. - * Returns 0 if the duration is unknown, and a negative value if the - * file is invalid. - */ -static float -faad_get_file_time_float(struct input_stream *is) -{ - struct decoder_buffer *buffer; - float length; - faacDecHandle decoder; - faacDecConfigurationPtr config; - - buffer = decoder_buffer_new(NULL, is, - FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); - length = faad_song_duration(buffer, is); - - if (length < 0) { - bool ret; - struct audio_format audio_format; - - decoder = faacDecOpen(); - - config = faacDecGetCurrentConfiguration(decoder); - config->outputFormat = FAAD_FMT_16BIT; - faacDecSetConfiguration(decoder, config); - - decoder_buffer_fill(buffer); - - ret = faad_decoder_init(decoder, buffer, &audio_format, NULL); - if (ret) - length = 0; - - faacDecClose(decoder); - } - - decoder_buffer_free(buffer); - - return length; -} - -/** - * Get a song file's total playing time in seconds, as an int. - * Returns 0 if the duration is unknown, and a negative value if the - * file is invalid. - */ -static int -faad_get_file_time(struct input_stream *is) -{ - int file_time = -1; - float length; - - if ((length = faad_get_file_time_float(is)) >= 0) - file_time = length + 0.5; - - return file_time; -} - -static void -faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) -{ - GError *error = NULL; - float total_time = 0; - faacDecHandle decoder; - struct audio_format audio_format; - faacDecConfigurationPtr config; - bool ret; - uint16_t bit_rate = 0; - struct decoder_buffer *buffer; - enum decoder_command cmd; - - buffer = decoder_buffer_new(mpd_decoder, is, - FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); - total_time = faad_song_duration(buffer, is); - - /* create the libfaad decoder */ - - decoder = faacDecOpen(); - - config = faacDecGetCurrentConfiguration(decoder); - config->outputFormat = FAAD_FMT_16BIT; -#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX - config->downMatrix = 1; -#endif -#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR - config->dontUpSampleImplicitSBR = 0; -#endif - faacDecSetConfiguration(decoder, config); - - while (!decoder_buffer_is_full(buffer) && - !input_stream_lock_eof(is) && - decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) { - adts_find_frame(buffer); - decoder_buffer_fill(buffer); - } - - /* initialize it */ - - ret = faad_decoder_init(decoder, buffer, &audio_format, &error); - if (!ret) { - g_warning("%s", error->message); - g_error_free(error); - faacDecClose(decoder); - return; - } - - /* initialize the MPD core */ - - decoder_initialized(mpd_decoder, &audio_format, false, total_time); - - /* the decoder loop */ - - do { - size_t frame_size; - const void *decoded; - faacDecFrameInfo frame_info; - - /* find the next frame */ - - frame_size = adts_find_frame(buffer); - if (frame_size == 0) - /* end of file */ - break; - - /* decode it */ - - decoded = faad_decoder_decode(decoder, buffer, &frame_info); - - if (frame_info.error > 0) { - g_warning("error decoding AAC stream: %s\n", - faacDecGetErrorMessage(frame_info.error)); - break; - } - - if (frame_info.channels != audio_format.channels) { - g_warning("channel count changed from %u to %u", - audio_format.channels, frame_info.channels); - break; - } - -#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE - if (frame_info.samplerate != audio_format.sample_rate) { - g_warning("sample rate changed from %u to %lu", - audio_format.sample_rate, - (unsigned long)frame_info.samplerate); - break; - } -#endif - - decoder_buffer_consume(buffer, frame_info.bytesconsumed); - - /* update bit rate and position */ - - if (frame_info.samples > 0) { - bit_rate = frame_info.bytesconsumed * 8.0 * - frame_info.channels * audio_format.sample_rate / - frame_info.samples / 1000 + 0.5; - } - - /* send PCM samples to MPD */ - - cmd = decoder_data(mpd_decoder, is, decoded, - (size_t)frame_info.samples * 2, - bit_rate); - } while (cmd != DECODE_COMMAND_STOP); - - /* cleanup */ - - faacDecClose(decoder); -} - -static bool -faad_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - int file_time = faad_get_file_time(is); - - if (file_time < 0) - return false; - - tag_handler_invoke_duration(handler, handler_ctx, file_time); - return true; -} - -static const char *const faad_suffixes[] = { "aac", NULL }; -static const char *const faad_mime_types[] = { - "audio/aac", "audio/aacp", NULL -}; - -const struct decoder_plugin faad_decoder_plugin = { - .name = "faad", - .stream_decode = faad_stream_decode, - .scan_stream = faad_scan_stream, - .suffixes = faad_suffixes, - .mime_types = faad_mime_types, -}; diff --git a/src/decoder/ffmpeg_decoder_plugin.c b/src/decoder/ffmpeg_decoder_plugin.c deleted file mode 100644 index 58bd2f54a..000000000 --- a/src/decoder/ffmpeg_decoder_plugin.c +++ /dev/null @@ -1,814 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "ffmpeg_metadata.h" -#include "tag_handler.h" - -#include <glib.h> - -#include <assert.h> -#include <stdio.h> -#include <unistd.h> -#include <stdlib.h> -#include <string.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <unistd.h> - -#include <libavcodec/avcodec.h> -#include <libavformat/avformat.h> -#include <libavformat/avio.h> -#include <libavutil/avutil.h> -#include <libavutil/log.h> -#include <libavutil/mathematics.h> -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,5,0) -#include <libavutil/dict.h> -#endif - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "ffmpeg" - -static GLogLevelFlags -level_ffmpeg_to_glib(int level) -{ - if (level <= AV_LOG_FATAL) - return G_LOG_LEVEL_CRITICAL; - - if (level <= AV_LOG_ERROR) - return G_LOG_LEVEL_WARNING; - - if (level <= AV_LOG_INFO) - return G_LOG_LEVEL_MESSAGE; - - return G_LOG_LEVEL_DEBUG; -} - -static void -mpd_ffmpeg_log_callback(G_GNUC_UNUSED void *ptr, int level, - const char *fmt, va_list vl) -{ - const AVClass * cls = NULL; - - if (ptr != NULL) - cls = *(const AVClass *const*)ptr; - - if (cls != NULL) { - char *domain = g_strconcat(G_LOG_DOMAIN, "/", cls->item_name(ptr), NULL); - g_logv(domain, level_ffmpeg_to_glib(level), fmt, vl); - g_free(domain); - } -} - -struct mpd_ffmpeg_stream { - struct decoder *decoder; - struct input_stream *input; - -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0) - AVIOContext *io; -#else - ByteIOContext *io; -#endif - unsigned char buffer[8192]; -}; - -static int -mpd_ffmpeg_stream_read(void *opaque, uint8_t *buf, int size) -{ - struct mpd_ffmpeg_stream *stream = opaque; - - return decoder_read(stream->decoder, stream->input, - (void *)buf, size); -} - -static int64_t -mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence) -{ - struct mpd_ffmpeg_stream *stream = opaque; - - if (whence == AVSEEK_SIZE) - return stream->input->size; - - if (!input_stream_lock_seek(stream->input, pos, whence, NULL)) - return -1; - - return stream->input->offset; -} - -static struct mpd_ffmpeg_stream * -mpd_ffmpeg_stream_open(struct decoder *decoder, struct input_stream *input) -{ - struct mpd_ffmpeg_stream *stream = g_new(struct mpd_ffmpeg_stream, 1); - stream->decoder = decoder; - stream->input = input; -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0) - stream->io = avio_alloc_context(stream->buffer, sizeof(stream->buffer), - false, stream, - mpd_ffmpeg_stream_read, NULL, - input->seekable - ? mpd_ffmpeg_stream_seek : NULL); -#else - stream->io = av_alloc_put_byte(stream->buffer, sizeof(stream->buffer), - false, stream, - mpd_ffmpeg_stream_read, NULL, - input->seekable - ? mpd_ffmpeg_stream_seek : NULL); -#endif - if (stream->io == NULL) { - g_free(stream); - return NULL; - } - - return stream; -} - -/** - * API compatibility wrapper for av_open_input_stream() and - * avformat_open_input(). - */ -static int -mpd_ffmpeg_open_input(AVFormatContext **ic_ptr, -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0) - AVIOContext *pb, -#else - ByteIOContext *pb, -#endif - const char *filename, - AVInputFormat *fmt) -{ -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,1,3) - AVFormatContext *context = avformat_alloc_context(); - if (context == NULL) - return AVERROR(ENOMEM); - - context->pb = pb; - *ic_ptr = context; - return avformat_open_input(ic_ptr, filename, fmt, NULL); -#else - return av_open_input_stream(ic_ptr, pb, filename, fmt, NULL); -#endif -} - -static void -mpd_ffmpeg_stream_close(struct mpd_ffmpeg_stream *stream) -{ - av_free(stream->io); - g_free(stream); -} - -static bool -ffmpeg_init(G_GNUC_UNUSED const struct config_param *param) -{ - av_log_set_callback(mpd_ffmpeg_log_callback); - - av_register_all(); - return true; -} - -static int -ffmpeg_find_audio_stream(const AVFormatContext *format_context) -{ - for (unsigned i = 0; i < format_context->nb_streams; ++i) - if (format_context->streams[i]->codec->codec_type == -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 64, 0) - AVMEDIA_TYPE_AUDIO) -#else - CODEC_TYPE_AUDIO) -#endif - return i; - - return -1; -} - -#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,25,0) -/** - * On some platforms, libavcodec wants the output buffer aligned to 16 - * bytes (because it uses SSE/Altivec internally). This function - * returns the aligned version of the specified buffer, and corrects - * the buffer size. - */ -static void * -align16(void *p, size_t *length_p) -{ - unsigned add = 16 - (size_t)p % 16; - - *length_p -= add; - return (char *)p + add; -} -#endif - -G_GNUC_CONST -static double -time_from_ffmpeg(int64_t t, const AVRational time_base) -{ - assert(t != (int64_t)AV_NOPTS_VALUE); - - return (double)av_rescale_q(t, time_base, (AVRational){1, 1024}) - / (double)1024; -} - -G_GNUC_CONST -static int64_t -time_to_ffmpeg(double t, const AVRational time_base) -{ - return av_rescale_q((int64_t)(t * 1024), (AVRational){1, 1024}, - time_base); -} - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0) - -static void -copy_interleave_frame2(uint8_t *dest, uint8_t **src, - unsigned nframes, unsigned nchannels, - unsigned sample_size) -{ - for (unsigned frame = 0; frame < nframes; ++frame) { - for (unsigned channel = 0; channel < nchannels; ++channel) { - memcpy(dest, src[channel] + frame * sample_size, - sample_size); - dest += sample_size; - } - } -} - -/** - * Copy PCM data from a AVFrame to an interleaved buffer. - */ -static int -copy_interleave_frame(const AVCodecContext *codec_context, - const AVFrame *frame, - uint8_t *buffer, size_t buffer_size) -{ - int plane_size; - const int data_size = - av_samples_get_buffer_size(&plane_size, - codec_context->channels, - frame->nb_samples, - codec_context->sample_fmt, 1); - if (buffer_size < (size_t)data_size) - /* buffer is too small - shouldn't happen */ - return AVERROR(EINVAL); - - if (av_sample_fmt_is_planar(codec_context->sample_fmt) && - codec_context->channels > 1) { - copy_interleave_frame2(buffer, frame->extended_data, - frame->nb_samples, - codec_context->channels, - av_get_bytes_per_sample(codec_context->sample_fmt)); - } else { - memcpy(buffer, frame->extended_data[0], data_size); - } - - return data_size; -} -#endif - -static enum decoder_command -ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, - const AVPacket *packet, - AVCodecContext *codec_context, - const AVRational *time_base) -{ - if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE) - decoder_timestamp(decoder, - time_from_ffmpeg(packet->pts, *time_base)); - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) - AVPacket packet2 = *packet; -#else - const uint8_t *packet_data = packet->data; - int packet_size = packet->size; -#endif - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0) - uint8_t aligned_buffer[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16]; - const size_t buffer_size = sizeof(aligned_buffer); -#else - /* libavcodec < 0.8 needs an aligned buffer */ - uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16]; - size_t buffer_size = sizeof(audio_buf); - int16_t *aligned_buffer = align16(audio_buf, &buffer_size); -#endif - - enum decoder_command cmd = DECODE_COMMAND_NONE; - while ( -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) - packet2.size > 0 && -#else - packet_size > 0 && -#endif - cmd == DECODE_COMMAND_NONE) { - int audio_size = buffer_size; -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0) - - AVFrame *frame = avcodec_alloc_frame(); - if (frame == NULL) { - g_warning("Could not allocate frame"); - break; - } - - int got_frame = 0; - int len = avcodec_decode_audio4(codec_context, - frame, &got_frame, - &packet2); - if (len >= 0 && got_frame) { - audio_size = copy_interleave_frame(codec_context, - frame, - aligned_buffer, - buffer_size); - if (audio_size < 0) - len = audio_size; - } else if (len >= 0) - len = -1; - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&frame); -#else - av_freep(&frame); -#endif - -#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) - int len = avcodec_decode_audio3(codec_context, - aligned_buffer, &audio_size, - &packet2); -#else - int len = avcodec_decode_audio2(codec_context, - aligned_buffer, &audio_size, - packet_data, packet_size); -#endif - - if (len < 0) { - /* if error, we skip the frame */ - g_message("decoding failed, frame skipped\n"); - break; - } - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) - packet2.data += len; - packet2.size -= len; -#else - packet_data += len; - packet_size -= len; -#endif - - if (audio_size <= 0) - continue; - - cmd = decoder_data(decoder, is, - aligned_buffer, audio_size, - codec_context->bit_rate / 1000); - } - return cmd; -} - -#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(52, 94, 1) -#define AVSampleFormat SampleFormat -#endif - -G_GNUC_CONST -static enum sample_format -ffmpeg_sample_format(enum AVSampleFormat sample_fmt) -{ - switch (sample_fmt) { -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1) - case AV_SAMPLE_FMT_S16: -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0) - case AV_SAMPLE_FMT_S16P: -#endif -#else - case SAMPLE_FMT_S16: -#endif - return SAMPLE_FORMAT_S16; - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1) - case AV_SAMPLE_FMT_S32: -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0) - case AV_SAMPLE_FMT_S32P: -#endif -#else - case SAMPLE_FMT_S32: -#endif - return SAMPLE_FORMAT_S32; - -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0) - case AV_SAMPLE_FMT_FLTP: - return SAMPLE_FORMAT_FLOAT; -#endif - - default: - break; - } - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1) - char buffer[64]; - const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer), - sample_fmt); - if (name != NULL) - g_warning("Unsupported libavcodec SampleFormat value: %s (%d)", - name, sample_fmt); - else -#endif - g_warning("Unsupported libavcodec SampleFormat value: %d", - sample_fmt); - return SAMPLE_FORMAT_UNDEFINED; -} - -static AVInputFormat * -ffmpeg_probe(struct decoder *decoder, struct input_stream *is) -{ - enum { - BUFFER_SIZE = 16384, - PADDING = 16, - }; - - unsigned char *buffer = g_malloc(BUFFER_SIZE); - size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE); - if (nbytes <= PADDING || - !input_stream_lock_seek(is, 0, SEEK_SET, NULL)) { - g_free(buffer); - return NULL; - } - - /* some ffmpeg parsers (e.g. ac3_parser.c) read a few bytes - beyond the declared buffer limit, which makes valgrind - angry; this workaround removes some padding from the buffer - size */ - nbytes -= PADDING; - - AVProbeData avpd = { - .buf = buffer, - .buf_size = nbytes, - .filename = is->uri, - }; - - AVInputFormat *format = av_probe_input_format(&avpd, true); - g_free(buffer); - - return format; -} - -static void -ffmpeg_decode(struct decoder *decoder, struct input_stream *input) -{ - AVInputFormat *input_format = ffmpeg_probe(decoder, input); - if (input_format == NULL) - return; - - g_debug("detected input format '%s' (%s)", - input_format->name, input_format->long_name); - - struct mpd_ffmpeg_stream *stream = - mpd_ffmpeg_stream_open(decoder, input); - if (stream == NULL) { - g_warning("Failed to open stream"); - return; - } - - //ffmpeg works with ours "fileops" helper - AVFormatContext *format_context = NULL; - if (mpd_ffmpeg_open_input(&format_context, stream->io, input->uri, - input_format) != 0) { - g_warning("Open failed\n"); - mpd_ffmpeg_stream_close(stream); - return; - } - -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,6,0) - const int find_result = - avformat_find_stream_info(format_context, NULL); -#else - const int find_result = av_find_stream_info(format_context); -#endif - if (find_result < 0) { - g_warning("Couldn't find stream info\n"); -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&format_context); -#else - av_close_input_stream(format_context); -#endif - mpd_ffmpeg_stream_close(stream); - return; - } - - int audio_stream = ffmpeg_find_audio_stream(format_context); - if (audio_stream == -1) { - g_warning("No audio stream inside\n"); -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&format_context); -#else - av_close_input_stream(format_context); -#endif - mpd_ffmpeg_stream_close(stream); - return; - } - - AVStream *av_stream = format_context->streams[audio_stream]; - - AVCodecContext *codec_context = av_stream->codec; - if (codec_context->codec_name[0] != 0) - g_debug("codec '%s'", codec_context->codec_name); - - AVCodec *codec = avcodec_find_decoder(codec_context->codec_id); - - if (!codec) { - g_warning("Unsupported audio codec\n"); -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&format_context); -#else - av_close_input_stream(format_context); -#endif - mpd_ffmpeg_stream_close(stream); - return; - } - - const enum sample_format sample_format = - ffmpeg_sample_format(codec_context->sample_fmt); - if (sample_format == SAMPLE_FORMAT_UNDEFINED) - return; - - GError *error = NULL; - struct audio_format audio_format; - if (!audio_format_init_checked(&audio_format, - codec_context->sample_rate, - sample_format, - codec_context->channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&format_context); -#else - av_close_input_stream(format_context); -#endif - mpd_ffmpeg_stream_close(stream); - return; - } - - /* the audio format must be read from AVCodecContext by now, - because avcodec_open() has been demonstrated to fill bogus - values into AVCodecContext.channels - a change that will be - reverted later by avcodec_decode_audio3() */ - -#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,6,0) - const int open_result = avcodec_open2(codec_context, codec, NULL); -#else - const int open_result = avcodec_open(codec_context, codec); -#endif - if (open_result < 0) { - g_warning("Could not open codec\n"); -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&format_context); -#else - av_close_input_stream(format_context); -#endif - mpd_ffmpeg_stream_close(stream); - return; - } - - int total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE - ? format_context->duration / AV_TIME_BASE - : 0; - - decoder_initialized(decoder, &audio_format, - input->seekable, total_time); - - enum decoder_command cmd; - do { - AVPacket packet; - if (av_read_frame(format_context, &packet) < 0) - /* end of file */ - break; - - if (packet.stream_index == audio_stream) - cmd = ffmpeg_send_packet(decoder, input, - &packet, codec_context, - &av_stream->time_base); - else - cmd = decoder_get_command(decoder); - - av_free_packet(&packet); - - if (cmd == DECODE_COMMAND_SEEK) { - int64_t where = - time_to_ffmpeg(decoder_seek_where(decoder), - av_stream->time_base); - - if (av_seek_frame(format_context, audio_stream, where, - AV_TIME_BASE) < 0) - decoder_seek_error(decoder); - else { - avcodec_flush_buffers(codec_context); - decoder_command_finished(decoder); - } - } - } while (cmd != DECODE_COMMAND_STOP); - - avcodec_close(codec_context); -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&format_context); -#else - av_close_input_stream(format_context); -#endif - mpd_ffmpeg_stream_close(stream); -} - -//no tag reading in ffmpeg, check if playable -static bool -ffmpeg_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - AVInputFormat *input_format = ffmpeg_probe(NULL, is); - if (input_format == NULL) - return false; - - struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(NULL, is); - if (stream == NULL) - return false; - - AVFormatContext *f = NULL; - if (mpd_ffmpeg_open_input(&f, stream->io, is->uri, - input_format) != 0) { - mpd_ffmpeg_stream_close(stream); - return false; - } - -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,6,0) - const int find_result = - avformat_find_stream_info(f, NULL); -#else - const int find_result = av_find_stream_info(f); -#endif - if (find_result < 0) { -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&f); -#else - av_close_input_stream(f); -#endif - mpd_ffmpeg_stream_close(stream); - return false; - } - - if (f->duration != (int64_t)AV_NOPTS_VALUE) - tag_handler_invoke_duration(handler, handler_ctx, - f->duration / AV_TIME_BASE); - -#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(52,101,0) - av_metadata_conv(f, NULL, f->iformat->metadata_conv); -#endif - - ffmpeg_scan_dictionary(f->metadata, handler, handler_ctx); - int idx = ffmpeg_find_audio_stream(f); - if (idx >= 0) - ffmpeg_scan_dictionary(f->streams[idx]->metadata, - handler, handler_ctx); - -#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0) - avformat_close_input(&f); -#else - av_close_input_stream(f); -#endif - mpd_ffmpeg_stream_close(stream); - - return true; -} - -/** - * A list of extensions found for the formats supported by ffmpeg. - * This list is current as of 02-23-09; To find out if there are more - * supported formats, check the ffmpeg changelog since this date for - * more formats. - */ -static const char *const ffmpeg_suffixes[] = { - "16sv", "3g2", "3gp", "4xm", "8svx", "aa3", "aac", "ac3", "afc", "aif", - "aifc", "aiff", "al", "alaw", "amr", "anim", "apc", "ape", "asf", - "atrac", "au", "aud", "avi", "avm2", "avs", "bap", "bfi", "c93", "cak", - "cin", "cmv", "cpk", "daud", "dct", "divx", "dts", "dv", "dvd", "dxa", - "eac3", "film", "flac", "flc", "fli", "fll", "flx", "flv", "g726", - "gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts", - "m4a", "m4b", "m4v", - "mad", - "mj2", "mjpeg", "mjpg", "mka", "mkv", "mlp", "mm", "mmf", "mov", "mp+", - "mp1", "mp2", "mp3", "mp4", "mpc", "mpeg", "mpg", "mpga", "mpp", "mpu", - "mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv", - "ogx", "oma", "ogg", "omg", "psp", "pva", "qcp", "qt", "r3d", "ra", - "ram", "rl2", "rm", "rmvb", "roq", "rpl", "rvc", "shn", "smk", "snd", - "sol", "son", "spx", "str", "swf", "tgi", "tgq", "tgv", "thp", "ts", - "tsp", "tta", "xa", "xvid", "uv", "uv2", "vb", "vid", "vob", "voc", - "vp6", "vmd", "wav", "webm", "wma", "wmv", "wsaud", "wsvga", "wv", - "wve", - NULL -}; - -static const char *const ffmpeg_mime_types[] = { - "application/flv", - "application/m4a", - "application/mp4", - "application/octet-stream", - "application/ogg", - "application/x-ms-wmz", - "application/x-ms-wmd", - "application/x-ogg", - "application/x-shockwave-flash", - "application/x-shorten", - "audio/8svx", - "audio/16sv", - "audio/aac", - "audio/ac3", - "audio/aiff" - "audio/amr", - "audio/basic", - "audio/flac", - "audio/m4a", - "audio/mp4", - "audio/mpeg", - "audio/musepack", - "audio/ogg", - "audio/qcelp", - "audio/vorbis", - "audio/vorbis+ogg", - "audio/x-8svx", - "audio/x-16sv", - "audio/x-aac", - "audio/x-ac3", - "audio/x-aiff" - "audio/x-alaw", - "audio/x-au", - "audio/x-dca", - "audio/x-eac3", - "audio/x-flac", - "audio/x-gsm", - "audio/x-mace", - "audio/x-matroska", - "audio/x-monkeys-audio", - "audio/x-mpeg", - "audio/x-ms-wma", - "audio/x-ms-wax", - "audio/x-musepack", - "audio/x-ogg", - "audio/x-vorbis", - "audio/x-vorbis+ogg", - "audio/x-pn-realaudio", - "audio/x-pn-multirate-realaudio", - "audio/x-speex", - "audio/x-tta" - "audio/x-voc", - "audio/x-wav", - "audio/x-wma", - "audio/x-wv", - "video/anim", - "video/quicktime", - "video/msvideo", - "video/ogg", - "video/theora", - "video/webm", - "video/x-dv", - "video/x-flv", - "video/x-matroska", - "video/x-mjpeg", - "video/x-mpeg", - "video/x-ms-asf", - "video/x-msvideo", - "video/x-ms-wmv", - "video/x-ms-wvx", - "video/x-ms-wm", - "video/x-ms-wmx", - "video/x-nut", - "video/x-pva", - "video/x-theora", - "video/x-vid", - "video/x-wmv", - "video/x-xvid", - - /* special value for the "ffmpeg" input plugin: all streams by - the "ffmpeg" input plugin shall be decoded by this - plugin */ - "audio/x-mpd-ffmpeg", - - NULL -}; - -const struct decoder_plugin ffmpeg_decoder_plugin = { - .name = "ffmpeg", - .init = ffmpeg_init, - .stream_decode = ffmpeg_decode, - .scan_stream = ffmpeg_scan_stream, - .suffixes = ffmpeg_suffixes, - .mime_types = ffmpeg_mime_types -}; diff --git a/src/decoder/ffmpeg_metadata.c b/src/decoder/ffmpeg_metadata.c deleted file mode 100644 index 3ef774f63..000000000 --- a/src/decoder/ffmpeg_metadata.c +++ /dev/null @@ -1,85 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "ffmpeg_metadata.h" -#include "tag_table.h" -#include "tag_handler.h" - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "ffmpeg" - -static const struct tag_table ffmpeg_tags[] = { -#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(52,50,0) - { "author", TAG_ARTIST }, -#endif - { "year", TAG_DATE }, - { "author-sort", TAG_ARTIST_SORT }, - { "album_artist", TAG_ALBUM_ARTIST }, - { "album_artist-sort", TAG_ALBUM_ARTIST_SORT }, - - /* sentinel */ - { NULL, TAG_NUM_OF_ITEM_TYPES } -}; - -static void -ffmpeg_copy_metadata(enum tag_type type, - AVDictionary *m, const char *name, - const struct tag_handler *handler, void *handler_ctx) -{ - AVDictionaryEntry *mt = NULL; - - while ((mt = av_dict_get(m, name, mt, 0)) != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - type, mt->value); -} - -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,5,0) - -static void -ffmpeg_scan_pairs(AVDictionary *dict, - const struct tag_handler *handler, void *handler_ctx) -{ - AVDictionaryEntry *i = NULL; - - while ((i = av_dict_get(dict, "", i, AV_DICT_IGNORE_SUFFIX)) != NULL) - tag_handler_invoke_pair(handler, handler_ctx, - i->key, i->value); -} - -#endif - -void -ffmpeg_scan_dictionary(AVDictionary *dict, - const struct tag_handler *handler, void *handler_ctx) -{ - for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) - ffmpeg_copy_metadata(i, dict, tag_item_names[i], - handler, handler_ctx); - - for (const struct tag_table *i = ffmpeg_tags; - i->name != NULL; ++i) - ffmpeg_copy_metadata(i->type, dict, i->name, - handler, handler_ctx); - -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,5,0) - if (handler->pair != NULL) - ffmpeg_scan_pairs(dict, handler, handler_ctx); -#endif -} diff --git a/src/decoder/ffmpeg_metadata.h b/src/decoder/ffmpeg_metadata.h deleted file mode 100644 index 60658f479..000000000 --- a/src/decoder/ffmpeg_metadata.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_FFMPEG_METADATA_H -#define MPD_FFMPEG_METADATA_H - -#include <libavformat/avformat.h> -#include <libavutil/avutil.h> -#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,5,0) -#include <libavutil/dict.h> -#endif - -#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(53,1,0) -#define AVDictionary AVMetadata -#define AVDictionaryEntry AVMetadataTag -#define av_dict_get av_metadata_get -#endif - -struct tag_handler; - -void -ffmpeg_scan_dictionary(AVDictionary *dict, - const struct tag_handler *handler, void *handler_ctx); - -#endif diff --git a/src/decoder/flac_compat.h b/src/decoder/flac_compat.h deleted file mode 100644 index 9a30acc26..000000000 --- a/src/decoder/flac_compat.h +++ /dev/null @@ -1,114 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* - * Common data structures and functions used by FLAC and OggFLAC - */ - -#ifndef MPD_FLAC_COMPAT_H -#define MPD_FLAC_COMPAT_H - -#include <FLAC/export.h> -#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 -# include <FLAC/seekable_stream_decoder.h> - -/* starting with libFLAC 1.1.3, the SeekableStreamDecoder has been - merged into the StreamDecoder. The following macros try to emulate - the new API for libFLAC 1.1.2 by mapping MPD's StreamDecoder calls - to the old SeekableStreamDecoder API. */ - -#define FLAC__StreamDecoder FLAC__SeekableStreamDecoder -#define FLAC__stream_decoder_new FLAC__seekable_stream_decoder_new -#define FLAC__stream_decoder_get_decode_position FLAC__seekable_stream_decoder_get_decode_position -#define FLAC__stream_decoder_get_state FLAC__seekable_stream_decoder_get_state -#define FLAC__stream_decoder_process_single FLAC__seekable_stream_decoder_process_single -#define FLAC__stream_decoder_process_until_end_of_metadata FLAC__seekable_stream_decoder_process_until_end_of_metadata -#define FLAC__stream_decoder_seek_absolute FLAC__seekable_stream_decoder_seek_absolute -#define FLAC__stream_decoder_finish FLAC__seekable_stream_decoder_finish -#define FLAC__stream_decoder_delete FLAC__seekable_stream_decoder_delete - -#define FLAC__STREAM_DECODER_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM - -typedef unsigned flac_read_status_size_t; - -#define FLAC__StreamDecoderReadStatus FLAC__SeekableStreamDecoderReadStatus -#define FLAC__STREAM_DECODER_READ_STATUS_CONTINUE FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK -#define FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK -#define FLAC__STREAM_DECODER_READ_STATUS_ABORT FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR - -#define FLAC__StreamDecoderSeekStatus FLAC__SeekableStreamDecoderSeekStatus -#define FLAC__STREAM_DECODER_SEEK_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK -#define FLAC__STREAM_DECODER_SEEK_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR -#define FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR - -#define FLAC__StreamDecoderTellStatus FLAC__SeekableStreamDecoderTellStatus -#define FLAC__STREAM_DECODER_TELL_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK -#define FLAC__STREAM_DECODER_TELL_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR -#define FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR - -#define FLAC__StreamDecoderLengthStatus FLAC__SeekableStreamDecoderLengthStatus -#define FLAC__STREAM_DECODER_LENGTH_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK -#define FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR -#define FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR - -typedef enum { - FLAC__STREAM_DECODER_INIT_STATUS_OK, - FLAC__STREAM_DECODER_INIT_STATUS_ERROR, -} FLAC__StreamDecoderInitStatus; - -static inline FLAC__StreamDecoderInitStatus -FLAC__stream_decoder_init_stream(FLAC__SeekableStreamDecoder *decoder, - FLAC__SeekableStreamDecoderReadCallback read_cb, - FLAC__SeekableStreamDecoderSeekCallback seek_cb, - FLAC__SeekableStreamDecoderTellCallback tell_cb, - FLAC__SeekableStreamDecoderLengthCallback length_cb, - FLAC__SeekableStreamDecoderEofCallback eof_cb, - FLAC__SeekableStreamDecoderWriteCallback write_cb, - FLAC__SeekableStreamDecoderMetadataCallback metadata_cb, - FLAC__SeekableStreamDecoderErrorCallback error_cb, - void *data) -{ - return FLAC__seekable_stream_decoder_set_read_callback(decoder, read_cb) && - FLAC__seekable_stream_decoder_set_seek_callback(decoder, seek_cb) && - FLAC__seekable_stream_decoder_set_tell_callback(decoder, tell_cb) && - FLAC__seekable_stream_decoder_set_length_callback(decoder, length_cb) && - FLAC__seekable_stream_decoder_set_eof_callback(decoder, eof_cb) && - FLAC__seekable_stream_decoder_set_write_callback(decoder, write_cb) && - FLAC__seekable_stream_decoder_set_metadata_callback(decoder, metadata_cb) && - FLAC__seekable_stream_decoder_set_metadata_respond(decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT) && - FLAC__seekable_stream_decoder_set_error_callback(decoder, error_cb) && - FLAC__seekable_stream_decoder_set_client_data(decoder, data) && - FLAC__seekable_stream_decoder_init(decoder) == FLAC__SEEKABLE_STREAM_DECODER_OK - ? FLAC__STREAM_DECODER_INIT_STATUS_OK - : FLAC__STREAM_DECODER_INIT_STATUS_ERROR; -} - -#else /* FLAC_API_VERSION_CURRENT > 7 */ - -# include <FLAC/stream_decoder.h> - -# define flac_init(a,b,c,d,e,f,g,h,i,j) \ - (FLAC__stream_decoder_init_stream(a,b,c,d,e,f,g,h,i,j) \ - == FLAC__STREAM_DECODER_INIT_STATUS_OK) - -typedef size_t flac_read_status_size_t; - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -#endif /* _FLAC_COMMON_H */ diff --git a/src/decoder/flac_decoder_plugin.c b/src/decoder/flac_decoder_plugin.c deleted file mode 100644 index fb0b3502d..000000000 --- a/src/decoder/flac_decoder_plugin.c +++ /dev/null @@ -1,486 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" /* must be first for large file support */ -#include "_flac_common.h" -#include "flac_compat.h" -#include "flac_metadata.h" - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 -#include "_ogg_common.h" -#endif - -#include <glib.h> - -#include <assert.h> -#include <unistd.h> - -#include <sys/stat.h> -#include <sys/types.h> - -/* this code was based on flac123, from flac-tools */ - -static FLAC__StreamDecoderReadStatus -flac_read_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, - FLAC__byte buf[], flac_read_status_size_t *bytes, - void *fdata) -{ - struct flac_data *data = fdata; - size_t r; - - r = decoder_read(data->decoder, data->input_stream, - (void *)buf, *bytes); - *bytes = r; - - if (r == 0) { - if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE || - input_stream_lock_eof(data->input_stream)) - return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM; - else - return FLAC__STREAM_DECODER_READ_STATUS_ABORT; - } - - return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; -} - -static FLAC__StreamDecoderSeekStatus -flac_seek_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, - FLAC__uint64 offset, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - if (!data->input_stream->seekable) - return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED; - - if (!input_stream_lock_seek(data->input_stream, offset, SEEK_SET, - NULL)) - return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR; - - return FLAC__STREAM_DECODER_SEEK_STATUS_OK; -} - -static FLAC__StreamDecoderTellStatus -flac_tell_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, - FLAC__uint64 * offset, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - if (!data->input_stream->seekable) - return FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED; - - *offset = (long)(data->input_stream->offset); - - return FLAC__STREAM_DECODER_TELL_STATUS_OK; -} - -static FLAC__StreamDecoderLengthStatus -flac_length_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, - FLAC__uint64 * length, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - if (data->input_stream->size < 0) - return FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED; - - *length = (size_t) (data->input_stream->size); - - return FLAC__STREAM_DECODER_LENGTH_STATUS_OK; -} - -static FLAC__bool -flac_eof_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE && - decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) || - input_stream_lock_eof(data->input_stream); -} - -static void -flac_error_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, - FLAC__StreamDecoderErrorStatus status, void *fdata) -{ - flac_error_common_cb(status, (struct flac_data *) fdata); -} - -#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 -static void flacPrintErroredState(FLAC__SeekableStreamDecoderState state) -{ - switch (state) { - case FLAC__SEEKABLE_STREAM_DECODER_OK: - case FLAC__SEEKABLE_STREAM_DECODER_SEEKING: - case FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM: - return; - - case FLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR: - case FLAC__SEEKABLE_STREAM_DECODER_READ_ERROR: - case FLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR: - case FLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR: - case FLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED: - case FLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK: - case FLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED: - break; - } - - g_warning("%s\n", FLAC__SeekableStreamDecoderStateString[state]); -} -#else /* FLAC_API_VERSION_CURRENT >= 7 */ -static void flacPrintErroredState(FLAC__StreamDecoderState state) -{ - switch (state) { - case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA: - case FLAC__STREAM_DECODER_READ_METADATA: - case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC: - case FLAC__STREAM_DECODER_READ_FRAME: - case FLAC__STREAM_DECODER_END_OF_STREAM: - return; - - case FLAC__STREAM_DECODER_OGG_ERROR: - case FLAC__STREAM_DECODER_SEEK_ERROR: - case FLAC__STREAM_DECODER_ABORTED: - case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR: - case FLAC__STREAM_DECODER_UNINITIALIZED: - break; - } - - g_warning("%s\n", FLAC__StreamDecoderStateString[state]); -} -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -static void flacMetadata(G_GNUC_UNUSED const FLAC__StreamDecoder * dec, - const FLAC__StreamMetadata * block, void *vdata) -{ - flac_metadata_common_cb(block, (struct flac_data *) vdata); -} - -static FLAC__StreamDecoderWriteStatus -flac_write_cb(const FLAC__StreamDecoder *dec, const FLAC__Frame *frame, - const FLAC__int32 *const buf[], void *vdata) -{ - struct flac_data *data = (struct flac_data *) vdata; - FLAC__uint64 nbytes = 0; - - if (FLAC__stream_decoder_get_decode_position(dec, &nbytes)) { - if (data->position > 0 && nbytes > data->position) { - nbytes -= data->position; - data->position += nbytes; - } else { - data->position = nbytes; - nbytes = 0; - } - } else - nbytes = 0; - - return flac_common_write(data, frame, buf, nbytes); -} - -static bool -flac_scan_file(const char *file, - const struct tag_handler *handler, void *handler_ctx) -{ - return flac_scan_file2(file, NULL, handler, handler_ctx); -} - -/** - * Some glue code around FLAC__stream_decoder_new(). - */ -static FLAC__StreamDecoder * -flac_decoder_new(void) -{ - FLAC__StreamDecoder *sd = FLAC__stream_decoder_new(); - if (sd == NULL) { - g_warning("FLAC__stream_decoder_new() failed"); - return NULL; - } - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - if(!FLAC__stream_decoder_set_metadata_respond(sd, FLAC__METADATA_TYPE_VORBIS_COMMENT)) - g_debug("FLAC__stream_decoder_set_metadata_respond() has failed"); -#endif - - return sd; -} - -static bool -flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd, - FLAC__uint64 duration) -{ - data->total_frames = duration; - - if (!FLAC__stream_decoder_process_until_end_of_metadata(sd)) { - g_warning("problem reading metadata"); - return false; - } - - if (data->initialized) { - /* done */ - decoder_initialized(data->decoder, &data->audio_format, - data->input_stream->seekable, - (float)data->total_frames / - (float)data->audio_format.sample_rate); - return true; - } - - if (data->input_stream->seekable) - /* allow the workaround below only for nonseekable - streams*/ - return false; - - /* no stream_info packet found; try to initialize the decoder - from the first frame header */ - FLAC__stream_decoder_process_single(sd); - return data->initialized; -} - -static void -flac_decoder_loop(struct flac_data *data, FLAC__StreamDecoder *flac_dec, - FLAC__uint64 t_start, FLAC__uint64 t_end) -{ - struct decoder *decoder = data->decoder; - enum decoder_command cmd; - - data->first_frame = t_start; - - while (true) { - if (data->tag != NULL && !tag_is_empty(data->tag)) { - cmd = decoder_tag(data->decoder, data->input_stream, - data->tag); - tag_free(data->tag); - data->tag = tag_new(); - } else - cmd = decoder_get_command(decoder); - - if (cmd == DECODE_COMMAND_SEEK) { - FLAC__uint64 seek_sample = t_start + - decoder_seek_where(decoder) * - data->audio_format.sample_rate; - if (seek_sample >= t_start && - (t_end == 0 || seek_sample <= t_end) && - FLAC__stream_decoder_seek_absolute(flac_dec, seek_sample)) { - data->next_frame = seek_sample; - data->position = 0; - decoder_command_finished(decoder); - } else - decoder_seek_error(decoder); - } else if (cmd == DECODE_COMMAND_STOP || - FLAC__stream_decoder_get_state(flac_dec) == FLAC__STREAM_DECODER_END_OF_STREAM) - break; - - if (t_end != 0 && data->next_frame >= t_end) - /* end of this sub track */ - break; - - if (!FLAC__stream_decoder_process_single(flac_dec) && - decoder_get_command(decoder) == DECODE_COMMAND_NONE) { - /* a failure that was not triggered by a - decoder command */ - flacPrintErroredState(FLAC__stream_decoder_get_state(flac_dec)); - break; - } - } -} - -static FLAC__StreamDecoderInitStatus -stream_init_oggflac(FLAC__StreamDecoder *flac_dec, struct flac_data *data) -{ -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - return FLAC__stream_decoder_init_ogg_stream(flac_dec, - flac_read_cb, - flac_seek_cb, - flac_tell_cb, - flac_length_cb, - flac_eof_cb, - flac_write_cb, - flacMetadata, - flac_error_cb, - data); -#else - (void)flac_dec; - (void)data; - - return FLAC__STREAM_DECODER_INIT_STATUS_ERROR; -#endif -} - -static FLAC__StreamDecoderInitStatus -stream_init_flac(FLAC__StreamDecoder *flac_dec, struct flac_data *data) -{ - return FLAC__stream_decoder_init_stream(flac_dec, - flac_read_cb, flac_seek_cb, - flac_tell_cb, flac_length_cb, - flac_eof_cb, flac_write_cb, - flacMetadata, - flac_error_cb, - data); -} - -static FLAC__StreamDecoderInitStatus -stream_init(FLAC__StreamDecoder *flac_dec, struct flac_data *data, bool is_ogg) -{ - return is_ogg - ? stream_init_oggflac(flac_dec, data) - : stream_init_flac(flac_dec, data); -} - -static void -flac_decode_internal(struct decoder * decoder, - struct input_stream *input_stream, - bool is_ogg) -{ - FLAC__StreamDecoder *flac_dec; - struct flac_data data; - - flac_dec = flac_decoder_new(); - if (flac_dec == NULL) - return; - - flac_data_init(&data, decoder, input_stream); - data.tag = tag_new(); - - FLAC__StreamDecoderInitStatus status = - stream_init(flac_dec, &data, is_ogg); - if (status != FLAC__STREAM_DECODER_INIT_STATUS_OK) { - flac_data_deinit(&data); - FLAC__stream_decoder_delete(flac_dec); -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - g_warning("%s", FLAC__StreamDecoderInitStatusString[status]); -#endif - return; - } - - if (!flac_decoder_initialize(&data, flac_dec, 0)) { - flac_data_deinit(&data); - FLAC__stream_decoder_finish(flac_dec); - FLAC__stream_decoder_delete(flac_dec); - return; - } - - flac_decoder_loop(&data, flac_dec, 0, 0); - - flac_data_deinit(&data); - - FLAC__stream_decoder_finish(flac_dec); - FLAC__stream_decoder_delete(flac_dec); -} - -static void -flac_decode(struct decoder * decoder, struct input_stream *input_stream) -{ - flac_decode_internal(decoder, input_stream, false); -} - -#ifndef HAVE_OGGFLAC - -static bool -oggflac_init(G_GNUC_UNUSED const struct config_param *param) -{ -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - return !!FLAC_API_SUPPORTS_OGG_FLAC; -#else - /* disable oggflac when libflac is too old */ - return false; -#endif -} - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - -static bool -oggflac_scan_file(const char *file, - const struct tag_handler *handler, void *handler_ctx) -{ - FLAC__Metadata_Iterator *it; - FLAC__StreamMetadata *block; - FLAC__Metadata_Chain *chain = FLAC__metadata_chain_new(); - - if (!(FLAC__metadata_chain_read_ogg(chain, file))) { - FLAC__metadata_chain_delete(chain); - return false; - } - - it = FLAC__metadata_iterator_new(); - FLAC__metadata_iterator_init(it, chain); - - do { - if (!(block = FLAC__metadata_iterator_get_block(it))) - break; - - flac_scan_metadata(NULL, block, - handler, handler_ctx); - } while (FLAC__metadata_iterator_next(it)); - FLAC__metadata_iterator_delete(it); - - FLAC__metadata_chain_delete(chain); - return true; -} - -static void -oggflac_decode(struct decoder *decoder, struct input_stream *input_stream) -{ - if (ogg_stream_type_detect(input_stream) != FLAC) - return; - - /* rewind the stream, because ogg_stream_type_detect() has - moved it */ - input_stream_lock_seek(input_stream, 0, SEEK_SET, NULL); - - flac_decode_internal(decoder, input_stream, true); -} - -static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL }; -static const char *const oggflac_mime_types[] = { - "application/ogg", - "application/x-ogg", - "audio/ogg", - "audio/x-flac+ogg", - "audio/x-ogg", - NULL -}; - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -const struct decoder_plugin oggflac_decoder_plugin = { - .name = "oggflac", - .init = oggflac_init, -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - .stream_decode = oggflac_decode, - .scan_file = oggflac_scan_file, - .suffixes = oggflac_suffixes, - .mime_types = oggflac_mime_types -#endif -}; - -#endif /* HAVE_OGGFLAC */ - -static const char *const flac_suffixes[] = { "flac", NULL }; -static const char *const flac_mime_types[] = { - "application/flac", - "application/x-flac", - "audio/flac", - "audio/x-flac", - NULL -}; - -const struct decoder_plugin flac_decoder_plugin = { - .name = "flac", - .stream_decode = flac_decode, - .scan_file = flac_scan_file, - .suffixes = flac_suffixes, - .mime_types = flac_mime_types, -}; diff --git a/src/decoder/flac_metadata.c b/src/decoder/flac_metadata.c deleted file mode 100644 index bd1eaf323..000000000 --- a/src/decoder/flac_metadata.c +++ /dev/null @@ -1,323 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "flac_metadata.h" -#include "replay_gain_info.h" -#include "tag.h" -#include "tag_handler.h" -#include "tag_table.h" - -#include <glib.h> - -#include <assert.h> -#include <stdbool.h> -#include <stdlib.h> - -static bool -flac_find_float_comment(const FLAC__StreamMetadata *block, - const char *cmnt, float *fl) -{ - int offset; - size_t pos; - int len; - unsigned char tmp, *p; - - offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, - cmnt); - if (offset < 0) - return false; - - pos = strlen(cmnt) + 1; /* 1 is for '=' */ - len = block->data.vorbis_comment.comments[offset].length - pos; - if (len <= 0) - return false; - - p = &block->data.vorbis_comment.comments[offset].entry[pos]; - tmp = p[len]; - p[len] = '\0'; - *fl = (float)atof((char *)p); - p[len] = tmp; - - return true; -} - -bool -flac_parse_replay_gain(struct replay_gain_info *rgi, - const FLAC__StreamMetadata *block) -{ - bool found = false; - - replay_gain_info_init(rgi); - - if (flac_find_float_comment(block, "replaygain_album_gain", - &rgi->tuples[REPLAY_GAIN_ALBUM].gain)) - found = true; - if (flac_find_float_comment(block, "replaygain_album_peak", - &rgi->tuples[REPLAY_GAIN_ALBUM].peak)) - found = true; - if (flac_find_float_comment(block, "replaygain_track_gain", - &rgi->tuples[REPLAY_GAIN_TRACK].gain)) - found = true; - if (flac_find_float_comment(block, "replaygain_track_peak", - &rgi->tuples[REPLAY_GAIN_TRACK].peak)) - found = true; - - return found; -} - -static bool -flac_find_string_comment(const FLAC__StreamMetadata *block, - const char *cmnt, char **str) -{ - int offset; - size_t pos; - int len; - const unsigned char *p; - - *str = NULL; - offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, - cmnt); - if (offset < 0) - return false; - - pos = strlen(cmnt) + 1; /* 1 is for '=' */ - len = block->data.vorbis_comment.comments[offset].length - pos; - if (len <= 0) - return false; - - p = &block->data.vorbis_comment.comments[offset].entry[pos]; - *str = g_strndup((const char *)p, len); - - return true; -} - -bool -flac_parse_mixramp(char **mixramp_start, char **mixramp_end, - const FLAC__StreamMetadata *block) -{ - bool found = false; - - if (flac_find_string_comment(block, "mixramp_start", mixramp_start)) - found = true; - if (flac_find_string_comment(block, "mixramp_end", mixramp_end)) - found = true; - - return found; -} - -/** - * Checks if the specified name matches the entry's name, and if yes, - * returns the comment value (not null-temrinated). - */ -static const char * -flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry, - const char *name, const char *char_tnum, size_t *length_r) -{ - size_t name_length = strlen(name); - size_t char_tnum_length = 0; - const char *comment = (const char*)entry->entry; - - if (entry->length <= name_length || - g_ascii_strncasecmp(comment, name, name_length) != 0) - return NULL; - - if (char_tnum != NULL) { - char_tnum_length = strlen(char_tnum); - if (entry->length > name_length + char_tnum_length + 2 && - comment[name_length] == '[' && - g_ascii_strncasecmp(comment + name_length + 1, - char_tnum, char_tnum_length) == 0 && - comment[name_length + char_tnum_length + 1] == ']') - name_length = name_length + char_tnum_length + 2; - else if (entry->length > name_length + char_tnum_length && - g_ascii_strncasecmp(comment + name_length, - char_tnum, char_tnum_length) == 0) - name_length = name_length + char_tnum_length; - } - - if (comment[name_length] == '=') { - *length_r = entry->length - name_length - 1; - return comment + name_length + 1; - } - - return NULL; -} - -/** - * Check if the comment's name equals the passed name, and if so, copy - * the comment value into the tag. - */ -static bool -flac_copy_comment(const FLAC__StreamMetadata_VorbisComment_Entry *entry, - const char *name, enum tag_type tag_type, - const char *char_tnum, - const struct tag_handler *handler, void *handler_ctx) -{ - const char *value; - size_t value_length; - - value = flac_comment_value(entry, name, char_tnum, &value_length); - if (value != NULL) { - char *p = g_strndup(value, value_length); - tag_handler_invoke_tag(handler, handler_ctx, tag_type, p); - g_free(p); - return true; - } - - return false; -} - -static const struct tag_table flac_tags[] = { - { "tracknumber", TAG_TRACK }, - { "discnumber", TAG_DISC }, - { "album artist", TAG_ALBUM_ARTIST }, - { NULL, TAG_NUM_OF_ITEM_TYPES } -}; - -static void -flac_scan_comment(const char *char_tnum, - const FLAC__StreamMetadata_VorbisComment_Entry *entry, - const struct tag_handler *handler, void *handler_ctx) -{ - if (handler->pair != NULL) { - char *name = g_strdup((const char*)entry->entry); - char *value = strchr(name, '='); - - if (value != NULL && value > name) { - *value++ = 0; - tag_handler_invoke_pair(handler, handler_ctx, - name, value); - } - - g_free(name); - } - - for (const struct tag_table *i = flac_tags; i->name != NULL; ++i) - if (flac_copy_comment(entry, i->name, i->type, char_tnum, - handler, handler_ctx)) - return; - - for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) - if (flac_copy_comment(entry, - tag_item_names[i], i, char_tnum, - handler, handler_ctx)) - return; -} - -static void -flac_scan_comments(const char *char_tnum, - const FLAC__StreamMetadata_VorbisComment *comment, - const struct tag_handler *handler, void *handler_ctx) -{ - for (unsigned i = 0; i < comment->num_comments; ++i) - flac_scan_comment(char_tnum, &comment->comments[i], - handler, handler_ctx); -} - -void -flac_scan_metadata(const char *track, - const FLAC__StreamMetadata *block, - const struct tag_handler *handler, void *handler_ctx) -{ - switch (block->type) { - case FLAC__METADATA_TYPE_VORBIS_COMMENT: - flac_scan_comments(track, &block->data.vorbis_comment, - handler, handler_ctx); - break; - - case FLAC__METADATA_TYPE_STREAMINFO: - if (block->data.stream_info.sample_rate > 0) - tag_handler_invoke_duration(handler, handler_ctx, - flac_duration(&block->data.stream_info)); - break; - - default: - break; - } -} - -void -flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum, - const FLAC__StreamMetadata_VorbisComment *comment) -{ - flac_scan_comments(char_tnum, comment, - &add_tag_handler, tag); -} - -bool -flac_scan_file2(const char *file, const char *char_tnum, - const struct tag_handler *handler, void *handler_ctx) -{ - FLAC__Metadata_SimpleIterator *it; - FLAC__StreamMetadata *block = NULL; - - it = FLAC__metadata_simple_iterator_new(); - if (!FLAC__metadata_simple_iterator_init(it, file, 1, 0)) { - const char *err; - FLAC_API FLAC__Metadata_SimpleIteratorStatus s; - - s = FLAC__metadata_simple_iterator_status(it); - - switch (s) { /* slightly more human-friendly messages: */ - case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ILLEGAL_INPUT: - err = "illegal input"; - break; - case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ERROR_OPENING_FILE: - err = "error opening file"; - break; - case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_NOT_A_FLAC_FILE: - err = "not a FLAC file"; - break; - default: - err = FLAC__Metadata_SimpleIteratorStatusString[s]; - } - g_debug("Reading '%s' metadata gave the following error: %s\n", - file, err); - FLAC__metadata_simple_iterator_delete(it); - return false; - } - - do { - block = FLAC__metadata_simple_iterator_get_block(it); - if (!block) - break; - - flac_scan_metadata(char_tnum, block, handler, handler_ctx); - FLAC__metadata_object_delete(block); - } while (FLAC__metadata_simple_iterator_next(it)); - - FLAC__metadata_simple_iterator_delete(it); - - return true; -} - -struct tag * -flac_tag_load(const char *file, const char *char_tnum) -{ - struct tag *tag = tag_new(); - - if (!flac_scan_file2(file, char_tnum, &add_tag_handler, tag) || - tag_is_empty(tag)) { - tag_free(tag); - tag = NULL; - } - - return tag; -} diff --git a/src/decoder/flac_metadata.h b/src/decoder/flac_metadata.h deleted file mode 100644 index 3c463d5d6..000000000 --- a/src/decoder/flac_metadata.h +++ /dev/null @@ -1,64 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_FLAC_METADATA_H -#define MPD_FLAC_METADATA_H - -#include <assert.h> -#include <stdbool.h> -#include <FLAC/metadata.h> - -struct tag_handler; -struct tag; -struct replay_gain_info; - -static inline unsigned -flac_duration(const FLAC__StreamMetadata_StreamInfo *stream_info) -{ - assert(stream_info->sample_rate > 0); - - return (stream_info->total_samples + stream_info->sample_rate - 1) / - stream_info->sample_rate; -} - -bool -flac_parse_replay_gain(struct replay_gain_info *rgi, - const FLAC__StreamMetadata *block); - -bool -flac_parse_mixramp(char **mixramp_start, char **mixramp_end, - const FLAC__StreamMetadata *block); - -void -flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum, - const FLAC__StreamMetadata_VorbisComment *comment); - -void -flac_scan_metadata(const char *track, - const FLAC__StreamMetadata *block, - const struct tag_handler *handler, void *handler_ctx); - -bool -flac_scan_file2(const char *file, const char *char_tnum, - const struct tag_handler *handler, void *handler_ctx); - -struct tag * -flac_tag_load(const char *file, const char *char_tnum); - -#endif diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c deleted file mode 100644 index 6964d8ac6..000000000 --- a/src/decoder/flac_pcm.c +++ /dev/null @@ -1,110 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "flac_pcm.h" - -#include <assert.h> - -static void flac_convert_stereo16(int16_t *dest, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - for (; position < end; ++position) { - *dest++ = buf[0][position]; - *dest++ = buf[1][position]; - } -} - -static void -flac_convert_16(int16_t *dest, - unsigned int num_channels, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - unsigned int c_chan; - - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} - -/** - * Note: this function also handles 24 bit files! - */ -static void -flac_convert_32(int32_t *dest, - unsigned int num_channels, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - unsigned int c_chan; - - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} - -static void -flac_convert_8(int8_t *dest, - unsigned int num_channels, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - unsigned int c_chan; - - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} - -void -flac_convert(void *dest, - unsigned int num_channels, enum sample_format sample_format, - const FLAC__int32 *const buf[], - unsigned int position, unsigned int end) -{ - switch (sample_format) { - case SAMPLE_FORMAT_S16: - if (num_channels == 2) - flac_convert_stereo16((int16_t*)dest, buf, - position, end); - else - flac_convert_16((int16_t*)dest, num_channels, buf, - position, end); - break; - - case SAMPLE_FORMAT_S24_P32: - case SAMPLE_FORMAT_S32: - flac_convert_32((int32_t*)dest, num_channels, buf, - position, end); - break; - - case SAMPLE_FORMAT_S8: - flac_convert_8((int8_t*)dest, num_channels, buf, - position, end); - break; - - case SAMPLE_FORMAT_FLOAT: - case SAMPLE_FORMAT_DSD: - case SAMPLE_FORMAT_UNDEFINED: - /* unreachable */ - assert(false); - } -} diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h deleted file mode 100644 index a931998c1..000000000 --- a/src/decoder/flac_pcm.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_FLAC_PCM_H -#define MPD_FLAC_PCM_H - -#include "audio_format.h" - -#include <FLAC/ordinals.h> - -void -flac_convert(void *dest, - unsigned int num_channels, enum sample_format sample_format, - const FLAC__int32 *const buf[], - unsigned int position, unsigned int end); - -#endif diff --git a/src/decoder/fluidsynth_decoder_plugin.c b/src/decoder/fluidsynth_decoder_plugin.c deleted file mode 100644 index 894b2d353..000000000 --- a/src/decoder/fluidsynth_decoder_plugin.c +++ /dev/null @@ -1,219 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "conf.h" - -#include <glib.h> - -#include <fluidsynth.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "fluidsynth" - -static unsigned sample_rate; -static const char *soundfont_path; - -/** - * Convert a fluidsynth log level to a GLib log level. - */ -static GLogLevelFlags -fluidsynth_level_to_glib(enum fluid_log_level level) -{ - switch (level) { - case FLUID_PANIC: - case FLUID_ERR: - return G_LOG_LEVEL_CRITICAL; - - case FLUID_WARN: - return G_LOG_LEVEL_WARNING; - - case FLUID_INFO: - return G_LOG_LEVEL_INFO; - - case FLUID_DBG: - case LAST_LOG_LEVEL: - return G_LOG_LEVEL_DEBUG; - } - - /* invalid fluidsynth log level */ - return G_LOG_LEVEL_MESSAGE; -} - -/** - * The fluidsynth logging callback. It forwards messages to the GLib - * logging library. - */ -static void -fluidsynth_mpd_log_function(int level, char *message, G_GNUC_UNUSED void *data) -{ - g_log(G_LOG_DOMAIN, fluidsynth_level_to_glib(level), "%s", message); -} - -static bool -fluidsynth_init(const struct config_param *param) -{ - GError *error = NULL; - - sample_rate = config_get_block_unsigned(param, "sample_rate", 48000); - if (!audio_check_sample_rate(sample_rate, &error)) { - g_warning("%s\n", error->message); - g_error_free(error); - return false; - } - - soundfont_path = - config_get_block_string(param, "soundfont", - "/usr/share/sounds/sf2/FluidR3_GM.sf2"); - - fluid_set_log_function(LAST_LOG_LEVEL, - fluidsynth_mpd_log_function, NULL); - - return true; -} - -static void -fluidsynth_file_decode(struct decoder *decoder, const char *path_fs) -{ - char setting_sample_rate[] = "synth.sample-rate"; - /* - char setting_verbose[] = "synth.verbose"; - char setting_yes[] = "yes"; - */ - fluid_settings_t *settings; - fluid_synth_t *synth; - fluid_player_t *player; - int ret; - enum decoder_command cmd; - - /* set up fluid settings */ - - settings = new_fluid_settings(); - if (settings == NULL) - return; - - fluid_settings_setnum(settings, setting_sample_rate, sample_rate); - - /* - fluid_settings_setstr(settings, setting_verbose, setting_yes); - */ - - /* create the fluid synth */ - - synth = new_fluid_synth(settings); - if (synth == NULL) { - delete_fluid_settings(settings); - return; - } - - ret = fluid_synth_sfload(synth, soundfont_path, true); - if (ret < 0) { - g_warning("fluid_synth_sfload() failed"); - delete_fluid_synth(synth); - delete_fluid_settings(settings); - return; - } - - /* create the fluid player */ - - player = new_fluid_player(synth); - if (player == NULL) { - delete_fluid_synth(synth); - delete_fluid_settings(settings); - return; - } - - ret = fluid_player_add(player, path_fs); - if (ret != 0) { - g_warning("fluid_player_add() failed"); - delete_fluid_player(player); - delete_fluid_synth(synth); - delete_fluid_settings(settings); - return; - } - - /* start the player */ - - ret = fluid_player_play(player); - if (ret != 0) { - g_warning("fluid_player_play() failed"); - delete_fluid_player(player); - delete_fluid_synth(synth); - delete_fluid_settings(settings); - return; - } - - /* initialization complete - announce the audio format to the - MPD core */ - - struct audio_format audio_format; - audio_format_init(&audio_format, sample_rate, SAMPLE_FORMAT_S16, 2); - decoder_initialized(decoder, &audio_format, false, -1); - - while (fluid_player_get_status(player) == FLUID_PLAYER_PLAYING) { - int16_t buffer[2048]; - const unsigned max_frames = G_N_ELEMENTS(buffer) / 2; - - /* read samples from fluidsynth and send them to the - MPD core */ - - ret = fluid_synth_write_s16(synth, max_frames, - buffer, 0, 2, - buffer, 1, 2); - if (ret != 0) - break; - - cmd = decoder_data(decoder, NULL, buffer, sizeof(buffer), - 0); - if (cmd != DECODE_COMMAND_NONE) - break; - } - - /* clean up */ - - fluid_player_stop(player); - fluid_player_join(player); - - delete_fluid_player(player); - delete_fluid_synth(synth); - delete_fluid_settings(settings); -} - -static bool -fluidsynth_scan_file(const char *file, - G_GNUC_UNUSED const struct tag_handler *handler, - G_GNUC_UNUSED void *handler_ctx) -{ - return fluid_is_midifile(file); -} - -static const char *const fluidsynth_suffixes[] = { - "mid", - NULL -}; - -const struct decoder_plugin fluidsynth_decoder_plugin = { - .name = "fluidsynth", - .init = fluidsynth_init, - .file_decode = fluidsynth_file_decode, - .scan_file = fluidsynth_scan_file, - .suffixes = fluidsynth_suffixes, -}; diff --git a/src/decoder/gme_decoder_plugin.c b/src/decoder/gme_decoder_plugin.c deleted file mode 100644 index 237a1deb1..000000000 --- a/src/decoder/gme_decoder_plugin.c +++ /dev/null @@ -1,257 +0,0 @@ -#include "config.h" -#include "../decoder_api.h" -#include "audio_check.h" -#include "uri.h" -#include "tag_handler.h" - -#include <glib.h> -#include <assert.h> -#include <errno.h> -#include <stdlib.h> -#include <string.h> - -#include <gme/gme.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "gme" - -#define SUBTUNE_PREFIX "tune_" - -enum { - GME_SAMPLE_RATE = 44100, - GME_CHANNELS = 2, - GME_BUFFER_FRAMES = 2048, - GME_BUFFER_SAMPLES = GME_BUFFER_FRAMES * GME_CHANNELS, -}; - -/** - * returns the file path stripped of any /tune_xxx.* subtune - * suffix - */ -static char * -get_container_name(const char *path_fs) -{ - const char *subtune_suffix = uri_get_suffix(path_fs); - char *path_container = g_strdup(path_fs); - char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", subtune_suffix, NULL); - GPatternSpec *path_with_subtune = g_pattern_spec_new(pat); - g_free(pat); - if (!g_pattern_match(path_with_subtune, - strlen(path_container), path_container, NULL)) { - g_pattern_spec_free(path_with_subtune); - return path_container; - } - - char *ptr = g_strrstr(path_container, "/" SUBTUNE_PREFIX); - if (ptr != NULL) - *ptr='\0'; - - g_pattern_spec_free(path_with_subtune); - return path_container; -} - -/** - * returns tune number from file.nsf/tune_xxx.* style path or 0 if no subtune - * is appended. - */ -static int -get_song_num(const char *path_fs) -{ - const char *subtune_suffix = uri_get_suffix(path_fs); - char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", subtune_suffix, NULL); - GPatternSpec *path_with_subtune = g_pattern_spec_new(pat); - g_free(pat); - - if (g_pattern_match(path_with_subtune, - strlen(path_fs), path_fs, NULL)) { - char *sub = g_strrstr(path_fs, "/" SUBTUNE_PREFIX); - g_pattern_spec_free(path_with_subtune); - if(!sub) - return 0; - - sub += strlen("/" SUBTUNE_PREFIX); - int song_num = strtol(sub, NULL, 10); - - return song_num - 1; - } else { - g_pattern_spec_free(path_with_subtune); - return 0; - } -} - -static char * -gme_container_scan(const char *path_fs, const unsigned int tnum) -{ - Music_Emu *emu; - const char* gme_err; - unsigned int num_songs; - - gme_err = gme_open_file(path_fs, &emu, GME_SAMPLE_RATE); - if (gme_err != NULL) { - g_warning("%s", gme_err); - return NULL; - } - - num_songs = gme_track_count(emu); - /* if it only contains a single tune, don't treat as container */ - if (num_songs < 2) - return NULL; - - const char *subtune_suffix = uri_get_suffix(path_fs); - if (tnum <= num_songs){ - char *subtune = g_strdup_printf( - SUBTUNE_PREFIX "%03u.%s", tnum, subtune_suffix); - return subtune; - } else - return NULL; -} - -static void -gme_file_decode(struct decoder *decoder, const char *path_fs) -{ - float song_len; - Music_Emu *emu; - gme_info_t *ti; - struct audio_format audio_format; - enum decoder_command cmd; - short buf[GME_BUFFER_SAMPLES]; - const char* gme_err; - char *path_container = get_container_name(path_fs); - int song_num = get_song_num(path_fs); - - gme_err = gme_open_file(path_container, &emu, GME_SAMPLE_RATE); - g_free(path_container); - if (gme_err != NULL) { - g_warning("%s", gme_err); - return; - } - - if((gme_err = gme_track_info(emu, &ti, song_num)) != NULL){ - g_warning("%s", gme_err); - gme_delete(emu); - return; - } - - if(ti->length > 0) - song_len = ti->length / 1000.0; - else song_len = -1; - - /* initialize the MPD decoder */ - - GError *error = NULL; - if (!audio_format_init_checked(&audio_format, GME_SAMPLE_RATE, - SAMPLE_FORMAT_S16, GME_CHANNELS, - &error)) { - g_warning("%s", error->message); - g_error_free(error); - gme_free_info(ti); - gme_delete(emu); - return; - } - - decoder_initialized(decoder, &audio_format, true, song_len); - - if((gme_err = gme_start_track(emu, song_num)) != NULL) - g_warning("%s", gme_err); - - if(ti->length > 0) - gme_set_fade(emu, ti->length); - - /* play */ - do { - gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf); - if (gme_err != NULL) { - g_warning("%s", gme_err); - return; - } - cmd = decoder_data(decoder, NULL, buf, sizeof(buf), 0); - - if(cmd == DECODE_COMMAND_SEEK) { - float where = decoder_seek_where(decoder); - if((gme_err = gme_seek(emu, (int)where*1000)) != NULL) - g_warning("%s", gme_err); - decoder_command_finished(decoder); - } - - if(gme_track_ended(emu)) - break; - } while(cmd != DECODE_COMMAND_STOP); - - gme_free_info(ti); - gme_delete(emu); -} - -static bool -gme_scan_file(const char *path_fs, - const struct tag_handler *handler, void *handler_ctx) -{ - Music_Emu *emu; - gme_info_t *ti; - const char* gme_err; - char *path_container=get_container_name(path_fs); - int song_num; - song_num=get_song_num(path_fs); - - gme_err = gme_open_file(path_container, &emu, GME_SAMPLE_RATE); - g_free(path_container); - if (gme_err != NULL) { - g_warning("%s", gme_err); - return false; - } - if((gme_err = gme_track_info(emu, &ti, song_num)) != NULL){ - g_warning("%s", gme_err); - gme_delete(emu); - return false; - } - - assert(ti != NULL); - - if(ti->length > 0) - tag_handler_invoke_duration(handler, handler_ctx, - ti->length / 100); - - if(ti->song != NULL){ - if(gme_track_count(emu) > 1){ - /* start numbering subtunes from 1 */ - char *tag_title=g_strdup_printf("%s (%d/%d)", - ti->song, song_num+1, gme_track_count(emu)); - tag_handler_invoke_tag(handler, handler_ctx, - TAG_TITLE, tag_title); - g_free(tag_title); - }else - tag_handler_invoke_tag(handler, handler_ctx, - TAG_TITLE, ti->song); - } - if(ti->author != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_ARTIST, ti->author); - if(ti->game != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_ALBUM, ti->game); - if(ti->comment != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_COMMENT, ti->comment); - if(ti->copyright != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_DATE, ti->copyright); - - gme_free_info(ti); - gme_delete(emu); - - return true; -} - -static const char *const gme_suffixes[] = { - "ay", "gbs", "gym", "hes", "kss", "nsf", - "nsfe", "sap", "spc", "vgm", "vgz", - NULL -}; - -extern const struct decoder_plugin gme_decoder_plugin; -const struct decoder_plugin gme_decoder_plugin = { - .name = "gme", - .file_decode = gme_file_decode, - .scan_file = gme_scan_file, - .suffixes = gme_suffixes, - .container_scan = gme_container_scan, -}; diff --git a/src/decoder/mad_decoder_plugin.c b/src/decoder/mad_decoder_plugin.c deleted file mode 100644 index 62c371642..000000000 --- a/src/decoder/mad_decoder_plugin.c +++ /dev/null @@ -1,1203 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "conf.h" -#include "tag_id3.h" -#include "tag_rva2.h" -#include "tag_handler.h" -#include "audio_check.h" - -#include <assert.h> -#include <unistd.h> -#include <stdlib.h> -#include <stdio.h> -#include <glib.h> -#include <mad.h> - -#ifdef HAVE_ID3TAG -#include <id3tag.h> -#endif - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "mad" - -#define FRAMES_CUSHION 2000 - -#define READ_BUFFER_SIZE 40960 - -enum mp3_action { - DECODE_SKIP = -3, - DECODE_BREAK = -2, - DECODE_CONT = -1, - DECODE_OK = 0 -}; - -enum muteframe { - MUTEFRAME_NONE, - MUTEFRAME_SKIP, - MUTEFRAME_SEEK -}; - -/* the number of samples of silence the decoder inserts at start */ -#define DECODERDELAY 529 - -#define DEFAULT_GAPLESS_MP3_PLAYBACK true - -static bool gapless_playback; - -static inline int32_t -mad_fixed_to_24_sample(mad_fixed_t sample) -{ - enum { - bits = 24, - MIN = -MAD_F_ONE, - MAX = MAD_F_ONE - 1 - }; - - /* round */ - sample = sample + (1L << (MAD_F_FRACBITS - bits)); - - /* clip */ - if (sample > MAX) - sample = MAX; - else if (sample < MIN) - sample = MIN; - - /* quantize */ - return sample >> (MAD_F_FRACBITS + 1 - bits); -} - -static void -mad_fixed_to_24_buffer(int32_t *dest, const struct mad_synth *synth, - unsigned int start, unsigned int end, - unsigned int num_channels) -{ - unsigned int i, c; - - for (i = start; i < end; ++i) { - for (c = 0; c < num_channels; ++c) - *dest++ = mad_fixed_to_24_sample(synth->pcm.samples[c][i]); - } -} - -static bool -mp3_plugin_init(G_GNUC_UNUSED const struct config_param *param) -{ - gapless_playback = config_get_bool(CONF_GAPLESS_MP3_PLAYBACK, - DEFAULT_GAPLESS_MP3_PLAYBACK); - return true; -} - -#define MP3_DATA_OUTPUT_BUFFER_SIZE 2048 - -struct mp3_data { - struct mad_stream stream; - struct mad_frame frame; - struct mad_synth synth; - mad_timer_t timer; - unsigned char input_buffer[READ_BUFFER_SIZE]; - int32_t output_buffer[MP3_DATA_OUTPUT_BUFFER_SIZE]; - float total_time; - float elapsed_time; - float seek_where; - enum muteframe mute_frame; - long *frame_offsets; - mad_timer_t *times; - unsigned long highest_frame; - unsigned long max_frames; - unsigned long current_frame; - unsigned int drop_start_frames; - unsigned int drop_end_frames; - unsigned int drop_start_samples; - unsigned int drop_end_samples; - bool found_replay_gain; - bool found_xing; - bool found_first_frame; - bool decoded_first_frame; - unsigned long bit_rate; - struct decoder *decoder; - struct input_stream *input_stream; - enum mad_layer layer; -}; - -static void -mp3_data_init(struct mp3_data *data, struct decoder *decoder, - struct input_stream *input_stream) -{ - data->mute_frame = MUTEFRAME_NONE; - data->highest_frame = 0; - data->max_frames = 0; - data->frame_offsets = NULL; - data->times = NULL; - data->current_frame = 0; - data->drop_start_frames = 0; - data->drop_end_frames = 0; - data->drop_start_samples = 0; - data->drop_end_samples = 0; - data->found_replay_gain = false; - data->found_xing = false; - data->found_first_frame = false; - data->decoded_first_frame = false; - data->decoder = decoder; - data->input_stream = input_stream; - data->layer = 0; - - mad_stream_init(&data->stream); - mad_stream_options(&data->stream, MAD_OPTION_IGNORECRC); - mad_frame_init(&data->frame); - mad_synth_init(&data->synth); - mad_timer_reset(&data->timer); -} - -static bool mp3_seek(struct mp3_data *data, long offset) -{ - if (!input_stream_lock_seek(data->input_stream, offset, SEEK_SET, NULL)) - return false; - - mad_stream_buffer(&data->stream, data->input_buffer, 0); - (data->stream).error = 0; - - return true; -} - -static bool -mp3_fill_buffer(struct mp3_data *data) -{ - size_t remaining, length; - unsigned char *dest; - - if (data->stream.next_frame != NULL) { - remaining = data->stream.bufend - data->stream.next_frame; - memmove(data->input_buffer, data->stream.next_frame, - remaining); - dest = (data->input_buffer) + remaining; - length = READ_BUFFER_SIZE - remaining; - } else { - remaining = 0; - length = READ_BUFFER_SIZE; - dest = data->input_buffer; - } - - /* we've exhausted the read buffer, so give up!, these potential - * mp3 frames are way too big, and thus unlikely to be mp3 frames */ - if (length == 0) - return false; - - length = decoder_read(data->decoder, data->input_stream, dest, length); - if (length == 0) - return false; - - mad_stream_buffer(&data->stream, data->input_buffer, - length + remaining); - (data->stream).error = 0; - - return true; -} - -#ifdef HAVE_ID3TAG -static bool -parse_id3_replay_gain_info(struct replay_gain_info *replay_gain_info, - struct id3_tag *tag) -{ - int i; - char *key; - char *value; - struct id3_frame *frame; - bool found = false; - - replay_gain_info_init(replay_gain_info); - - for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) { - if (frame->nfields < 3) - continue; - - key = (char *) - id3_ucs4_latin1duplicate(id3_field_getstring - (&frame->fields[1])); - value = (char *) - id3_ucs4_latin1duplicate(id3_field_getstring - (&frame->fields[2])); - - if (g_ascii_strcasecmp(key, "replaygain_track_gain") == 0) { - replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = atof(value); - found = true; - } else if (g_ascii_strcasecmp(key, "replaygain_album_gain") == 0) { - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = atof(value); - found = true; - } else if (g_ascii_strcasecmp(key, "replaygain_track_peak") == 0) { - replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = atof(value); - found = true; - } else if (g_ascii_strcasecmp(key, "replaygain_album_peak") == 0) { - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = atof(value); - found = true; - } - - free(key); - free(value); - } - - return found || - /* fall back on RVA2 if no replaygain tags found */ - tag_rva2_parse(tag, replay_gain_info); -} -#endif - -#ifdef HAVE_ID3TAG -static bool -parse_id3_mixramp(char **mixramp_start, char **mixramp_end, - struct id3_tag *tag) -{ - int i; - char *key; - char *value; - struct id3_frame *frame; - bool found = false; - - *mixramp_start = NULL; - *mixramp_end = NULL; - - for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) { - if (frame->nfields < 3) - continue; - - key = (char *) - id3_ucs4_latin1duplicate(id3_field_getstring - (&frame->fields[1])); - value = (char *) - id3_ucs4_latin1duplicate(id3_field_getstring - (&frame->fields[2])); - - if (g_ascii_strcasecmp(key, "mixramp_start") == 0) { - *mixramp_start = g_strdup(value); - found = true; - } else if (g_ascii_strcasecmp(key, "mixramp_end") == 0) { - *mixramp_end = g_strdup(value); - found = true; - } - - free(key); - free(value); - } - - return found; -} -#endif - -static void mp3_parse_id3(struct mp3_data *data, size_t tagsize, - struct tag **mpd_tag) -{ -#ifdef HAVE_ID3TAG - struct id3_tag *id3_tag = NULL; - id3_length_t count; - id3_byte_t const *id3_data; - id3_byte_t *allocated = NULL; - - count = data->stream.bufend - data->stream.this_frame; - - if (tagsize <= count) { - id3_data = data->stream.this_frame; - mad_stream_skip(&(data->stream), tagsize); - } else { - allocated = g_malloc(tagsize); - memcpy(allocated, data->stream.this_frame, count); - mad_stream_skip(&(data->stream), count); - - while (count < tagsize) { - size_t len; - - len = decoder_read(data->decoder, data->input_stream, - allocated + count, tagsize - count); - if (len == 0) - break; - else - count += len; - } - - if (count != tagsize) { - g_debug("error parsing ID3 tag"); - g_free(allocated); - return; - } - - id3_data = allocated; - } - - id3_tag = id3_tag_parse(id3_data, tagsize); - if (id3_tag == NULL) { - g_free(allocated); - return; - } - - if (mpd_tag) { - struct tag *tmp_tag = tag_id3_import(id3_tag); - if (tmp_tag != NULL) { - if (*mpd_tag != NULL) - tag_free(*mpd_tag); - *mpd_tag = tmp_tag; - } - } - - if (data->decoder != NULL) { - struct replay_gain_info rgi; - char *mixramp_start; - char *mixramp_end; - float replay_gain_db = 0; - - if (parse_id3_replay_gain_info(&rgi, id3_tag)) { - replay_gain_db = decoder_replay_gain(data->decoder, &rgi); - data->found_replay_gain = true; - } - - if (parse_id3_mixramp(&mixramp_start, &mixramp_end, id3_tag)) - decoder_mixramp(data->decoder, replay_gain_db, - mixramp_start, mixramp_end); - } - - id3_tag_delete(id3_tag); - - g_free(allocated); -#else /* !HAVE_ID3TAG */ - (void)mpd_tag; - - /* This code is enabled when libid3tag is disabled. Instead - of parsing the ID3 frame, it just skips it. */ - - size_t count = data->stream.bufend - data->stream.this_frame; - - if (tagsize <= count) { - mad_stream_skip(&data->stream, tagsize); - } else { - mad_stream_skip(&data->stream, count); - - while (count < tagsize) { - size_t len = tagsize - count; - char ignored[1024]; - if (len > sizeof(ignored)) - len = sizeof(ignored); - - len = decoder_read(data->decoder, data->input_stream, - ignored, len); - if (len == 0) - break; - else - count += len; - } - } -#endif -} - -#ifndef HAVE_ID3TAG -/** - * This function emulates libid3tag when it is disabled. Instead of - * doing a real analyzation of the frame, it just checks whether the - * frame begins with the string "ID3". If so, it returns the length - * of the ID3 frame. - */ -static signed long -id3_tag_query(const void *p0, size_t length) -{ - const char *p = p0; - - return length >= 10 && memcmp(p, "ID3", 3) == 0 - ? (p[8] << 7) + p[9] + 10 - : 0; -} -#endif /* !HAVE_ID3TAG */ - -static enum mp3_action -decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag) -{ - enum mad_layer layer; - - if ((data->stream).buffer == NULL - || (data->stream).error == MAD_ERROR_BUFLEN) { - if (!mp3_fill_buffer(data)) - return DECODE_BREAK; - } - if (mad_header_decode(&data->frame.header, &data->stream)) { - if ((data->stream).error == MAD_ERROR_LOSTSYNC && - (data->stream).this_frame) { - signed long tagsize = id3_tag_query((data->stream). - this_frame, - (data->stream). - bufend - - (data->stream). - this_frame); - - if (tagsize > 0) { - if (tag && !(*tag)) { - mp3_parse_id3(data, (size_t)tagsize, - tag); - } else { - mad_stream_skip(&(data->stream), - tagsize); - } - return DECODE_CONT; - } - } - if (MAD_RECOVERABLE((data->stream).error)) { - return DECODE_SKIP; - } else { - if ((data->stream).error == MAD_ERROR_BUFLEN) - return DECODE_CONT; - else { - g_warning("unrecoverable frame level error " - "(%s).\n", - mad_stream_errorstr(&data->stream)); - return DECODE_BREAK; - } - } - } - - layer = data->frame.header.layer; - if (!data->layer) { - if (layer != MAD_LAYER_II && layer != MAD_LAYER_III) { - /* Only layer 2 and 3 have been tested to work */ - return DECODE_SKIP; - } - data->layer = layer; - } else if (layer != data->layer) { - /* Don't decode frames with a different layer than the first */ - return DECODE_SKIP; - } - - return DECODE_OK; -} - -static enum mp3_action -decodeNextFrame(struct mp3_data *data) -{ - if ((data->stream).buffer == NULL - || (data->stream).error == MAD_ERROR_BUFLEN) { - if (!mp3_fill_buffer(data)) - return DECODE_BREAK; - } - if (mad_frame_decode(&data->frame, &data->stream)) { - if ((data->stream).error == MAD_ERROR_LOSTSYNC) { - signed long tagsize = id3_tag_query((data->stream). - this_frame, - (data->stream). - bufend - - (data->stream). - this_frame); - if (tagsize > 0) { - mad_stream_skip(&(data->stream), tagsize); - return DECODE_CONT; - } - } - if (MAD_RECOVERABLE((data->stream).error)) { - return DECODE_SKIP; - } else { - if ((data->stream).error == MAD_ERROR_BUFLEN) - return DECODE_CONT; - else { - g_warning("unrecoverable frame level error " - "(%s).\n", - mad_stream_errorstr(&data->stream)); - return DECODE_BREAK; - } - } - } - - return DECODE_OK; -} - -/* xing stuff stolen from alsaplayer, and heavily modified by jat */ -#define XI_MAGIC (('X' << 8) | 'i') -#define NG_MAGIC (('n' << 8) | 'g') -#define IN_MAGIC (('I' << 8) | 'n') -#define FO_MAGIC (('f' << 8) | 'o') - -enum xing_magic { - XING_MAGIC_XING, /* VBR */ - XING_MAGIC_INFO /* CBR */ -}; - -struct xing { - long flags; /* valid fields (see below) */ - unsigned long frames; /* total number of frames */ - unsigned long bytes; /* total number of bytes */ - unsigned char toc[100]; /* 100-point seek table */ - long scale; /* VBR quality */ - enum xing_magic magic; /* header magic */ -}; - -enum { - XING_FRAMES = 0x00000001L, - XING_BYTES = 0x00000002L, - XING_TOC = 0x00000004L, - XING_SCALE = 0x00000008L -}; - -struct lame_version { - unsigned major; - unsigned minor; -}; - -struct lame { - char encoder[10]; /* 9 byte encoder name/version ("LAME3.97b") */ - struct lame_version version; /* struct containing just the version */ - float peak; /* replaygain peak */ - float track_gain; /* replaygain track gain */ - float album_gain; /* replaygain album gain */ - int encoder_delay; /* # of added samples at start of mp3 */ - int encoder_padding; /* # of added samples at end of mp3 */ - int crc; /* CRC of the first 190 bytes of this frame */ -}; - -static bool -parse_xing(struct xing *xing, struct mad_bitptr *ptr, int *oldbitlen) -{ - unsigned long bits; - int bitlen; - int bitsleft; - int i; - - bitlen = *oldbitlen; - - if (bitlen < 16) - return false; - - bits = mad_bit_read(ptr, 16); - bitlen -= 16; - - if (bits == XI_MAGIC) { - if (bitlen < 16) - return false; - - if (mad_bit_read(ptr, 16) != NG_MAGIC) - return false; - - bitlen -= 16; - xing->magic = XING_MAGIC_XING; - } else if (bits == IN_MAGIC) { - if (bitlen < 16) - return false; - - if (mad_bit_read(ptr, 16) != FO_MAGIC) - return false; - - bitlen -= 16; - xing->magic = XING_MAGIC_INFO; - } - else if (bits == NG_MAGIC) xing->magic = XING_MAGIC_XING; - else if (bits == FO_MAGIC) xing->magic = XING_MAGIC_INFO; - else - return false; - - if (bitlen < 32) - return false; - xing->flags = mad_bit_read(ptr, 32); - bitlen -= 32; - - if (xing->flags & XING_FRAMES) { - if (bitlen < 32) - return false; - xing->frames = mad_bit_read(ptr, 32); - bitlen -= 32; - } - - if (xing->flags & XING_BYTES) { - if (bitlen < 32) - return false; - xing->bytes = mad_bit_read(ptr, 32); - bitlen -= 32; - } - - if (xing->flags & XING_TOC) { - if (bitlen < 800) - return false; - for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(ptr, 8); - bitlen -= 800; - } - - if (xing->flags & XING_SCALE) { - if (bitlen < 32) - return false; - xing->scale = mad_bit_read(ptr, 32); - bitlen -= 32; - } - - /* Make sure we consume no less than 120 bytes (960 bits) in hopes that - * the LAME tag is found there, and not right after the Xing header */ - bitsleft = 960 - ((*oldbitlen) - bitlen); - if (bitsleft < 0) - return false; - else if (bitsleft > 0) { - mad_bit_read(ptr, bitsleft); - bitlen -= bitsleft; - } - - *oldbitlen = bitlen; - - return true; -} - -static bool -parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen) -{ - int adj = 0; - int name; - int orig; - int sign; - int gain; - int i; - - /* Unlike the xing header, the lame tag has a fixed length. Fail if - * not all 36 bytes (288 bits) are there. */ - if (*bitlen < 288) - return false; - - for (i = 0; i < 9; i++) - lame->encoder[i] = (char)mad_bit_read(ptr, 8); - lame->encoder[9] = '\0'; - - *bitlen -= 72; - - /* This is technically incorrect, since the encoder might not be lame. - * But there's no other way to determine if this is a lame tag, and we - * wouldn't want to go reading a tag that's not there. */ - if (!g_str_has_prefix(lame->encoder, "LAME")) - return false; - - if (sscanf(lame->encoder+4, "%u.%u", - &lame->version.major, &lame->version.minor) != 2) - return false; - - g_debug("detected LAME version %i.%i (\"%s\")\n", - lame->version.major, lame->version.minor, lame->encoder); - - /* The reference volume was changed from the 83dB used in the - * ReplayGain spec to 89dB in lame 3.95.1. Bump the gain for older - * versions, since everyone else uses 89dB instead of 83dB. - * Unfortunately, lame didn't differentiate between 3.95 and 3.95.1, so - * it's impossible to make the proper adjustment for 3.95. - * Fortunately, 3.95 was only out for about a day before 3.95.1 was - * released. -- tmz */ - if (lame->version.major < 3 || - (lame->version.major == 3 && lame->version.minor < 95)) - adj = 6; - - mad_bit_read(ptr, 16); - - lame->peak = mad_f_todouble(mad_bit_read(ptr, 32) << 5); /* peak */ - g_debug("LAME peak found: %f\n", lame->peak); - - lame->track_gain = 0; - name = mad_bit_read(ptr, 3); /* gain name */ - orig = mad_bit_read(ptr, 3); /* gain originator */ - sign = mad_bit_read(ptr, 1); /* sign bit */ - gain = mad_bit_read(ptr, 9); /* gain*10 */ - if (gain && name == 1 && orig != 0) { - lame->track_gain = ((sign ? -gain : gain) / 10.0) + adj; - g_debug("LAME track gain found: %f\n", lame->track_gain); - } - - /* tmz reports that this isn't currently written by any version of lame - * (as of 3.97). Since we have no way of testing it, don't use it. - * Wouldn't want to go blowing someone's ears just because we read it - * wrong. :P -- jat */ - lame->album_gain = 0; -#if 0 - name = mad_bit_read(ptr, 3); /* gain name */ - orig = mad_bit_read(ptr, 3); /* gain originator */ - sign = mad_bit_read(ptr, 1); /* sign bit */ - gain = mad_bit_read(ptr, 9); /* gain*10 */ - if (gain && name == 2 && orig != 0) { - lame->album_gain = ((sign ? -gain : gain) / 10.0) + adj; - g_debug("LAME album gain found: %f\n", lame->track_gain); - } -#else - mad_bit_read(ptr, 16); -#endif - - mad_bit_read(ptr, 16); - - lame->encoder_delay = mad_bit_read(ptr, 12); - lame->encoder_padding = mad_bit_read(ptr, 12); - - g_debug("encoder delay is %i, encoder padding is %i\n", - lame->encoder_delay, lame->encoder_padding); - - mad_bit_read(ptr, 80); - - lame->crc = mad_bit_read(ptr, 16); - - *bitlen -= 216; - - return true; -} - -static inline float -mp3_frame_duration(const struct mad_frame *frame) -{ - return mad_timer_count(frame->header.duration, - MAD_UNITS_MILLISECONDS) / 1000.0; -} - -static goffset -mp3_this_frame_offset(const struct mp3_data *data) -{ - goffset offset = data->input_stream->offset; - - if (data->stream.this_frame != NULL) - offset -= data->stream.bufend - data->stream.this_frame; - else - offset -= data->stream.bufend - data->stream.buffer; - - return offset; -} - -static goffset -mp3_rest_including_this_frame(const struct mp3_data *data) -{ - return data->input_stream->size - mp3_this_frame_offset(data); -} - -/** - * Attempt to calulcate the length of the song from filesize - */ -static void -mp3_filesize_to_song_length(struct mp3_data *data) -{ - goffset rest = mp3_rest_including_this_frame(data); - - if (rest > 0) { - float frame_duration = mp3_frame_duration(&data->frame); - - data->total_time = (rest * 8.0) / (data->frame).header.bitrate; - data->max_frames = data->total_time / frame_duration + - FRAMES_CUSHION; - } else { - data->max_frames = FRAMES_CUSHION; - data->total_time = 0; - } -} - -static bool -mp3_decode_first_frame(struct mp3_data *data, struct tag **tag) -{ - struct xing xing; - struct lame lame; - struct mad_bitptr ptr; - int bitlen; - enum mp3_action ret; - - /* stfu gcc */ - memset(&xing, 0, sizeof(struct xing)); - xing.flags = 0; - - while (true) { - do { - ret = decode_next_frame_header(data, tag); - } while (ret == DECODE_CONT); - if (ret == DECODE_BREAK) - return false; - if (ret == DECODE_SKIP) continue; - - do { - ret = decodeNextFrame(data); - } while (ret == DECODE_CONT); - if (ret == DECODE_BREAK) - return false; - if (ret == DECODE_OK) break; - } - - ptr = data->stream.anc_ptr; - bitlen = data->stream.anc_bitlen; - - mp3_filesize_to_song_length(data); - - /* - * if an xing tag exists, use that! - */ - if (parse_xing(&xing, &ptr, &bitlen)) { - data->found_xing = true; - data->mute_frame = MUTEFRAME_SKIP; - - if ((xing.flags & XING_FRAMES) && xing.frames) { - mad_timer_t duration = data->frame.header.duration; - mad_timer_multiply(&duration, xing.frames); - data->total_time = ((float)mad_timer_count(duration, MAD_UNITS_MILLISECONDS)) / 1000; - data->max_frames = xing.frames; - } - - if (parse_lame(&lame, &ptr, &bitlen)) { - if (gapless_playback && - data->input_stream->seekable) { - data->drop_start_samples = lame.encoder_delay + - DECODERDELAY; - data->drop_end_samples = lame.encoder_padding; - } - - /* Album gain isn't currently used. See comment in - * parse_lame() for details. -- jat */ - if (data->decoder != NULL && - !data->found_replay_gain && - lame.track_gain) { - struct replay_gain_info rgi; - replay_gain_info_init(&rgi); - rgi.tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain; - rgi.tuples[REPLAY_GAIN_TRACK].peak = lame.peak; - decoder_replay_gain(data->decoder, &rgi); - } - } - } - - if (!data->max_frames) - return false; - - if (data->max_frames > 8 * 1024 * 1024) { - g_warning("mp3 file header indicates too many frames: %lu\n", - data->max_frames); - return false; - } - - data->frame_offsets = g_malloc(sizeof(long) * data->max_frames); - data->times = g_malloc(sizeof(mad_timer_t) * data->max_frames); - - return true; -} - -static void mp3_data_finish(struct mp3_data *data) -{ - mad_synth_finish(&data->synth); - mad_frame_finish(&data->frame); - mad_stream_finish(&data->stream); - - g_free(data->frame_offsets); - g_free(data->times); -} - -/* this is primarily used for getting total time for tags */ -static int -mad_decoder_total_file_time(struct input_stream *is) -{ - struct mp3_data data; - int ret; - - mp3_data_init(&data, NULL, is); - if (!mp3_decode_first_frame(&data, NULL)) - ret = -1; - else - ret = data.total_time + 0.5; - mp3_data_finish(&data); - - return ret; -} - -static bool -mp3_open(struct input_stream *is, struct mp3_data *data, - struct decoder *decoder, struct tag **tag) -{ - mp3_data_init(data, decoder, is); - *tag = NULL; - if (!mp3_decode_first_frame(data, tag)) { - mp3_data_finish(data); - if (tag && *tag) - tag_free(*tag); - return false; - } - - return true; -} - -static long -mp3_time_to_frame(const struct mp3_data *data, double t) -{ - unsigned long i; - - for (i = 0; i < data->highest_frame; ++i) { - double frame_time = - mad_timer_count(data->times[i], - MAD_UNITS_MILLISECONDS) / 1000.; - if (frame_time >= t) - break; - } - - return i; -} - -static void -mp3_update_timer_next_frame(struct mp3_data *data) -{ - if (data->current_frame >= data->highest_frame) { - /* record this frame's properties in - data->frame_offsets (for seeking) and - data->times */ - data->bit_rate = (data->frame).header.bitrate; - - if (data->current_frame >= data->max_frames) - /* cap data->current_frame */ - data->current_frame = data->max_frames - 1; - else - data->highest_frame++; - - data->frame_offsets[data->current_frame] = - mp3_this_frame_offset(data); - - mad_timer_add(&data->timer, (data->frame).header.duration); - data->times[data->current_frame] = data->timer; - } else - /* get the new timer value from data->times */ - data->timer = data->times[data->current_frame]; - - data->current_frame++; - data->elapsed_time = - mad_timer_count(data->timer, MAD_UNITS_MILLISECONDS) / 1000.0; -} - -/** - * Sends the synthesized current frame via decoder_data(). - */ -static enum decoder_command -mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length) -{ - unsigned max_samples; - - max_samples = sizeof(data->output_buffer) / - sizeof(data->output_buffer[0]) / - MAD_NCHANNELS(&(data->frame).header); - - while (i < pcm_length) { - enum decoder_command cmd; - unsigned int num_samples = pcm_length - i; - if (num_samples > max_samples) - num_samples = max_samples; - - i += num_samples; - - mad_fixed_to_24_buffer(data->output_buffer, - &data->synth, - i - num_samples, i, - MAD_NCHANNELS(&(data->frame).header)); - num_samples *= MAD_NCHANNELS(&(data->frame).header); - - cmd = decoder_data(data->decoder, data->input_stream, - data->output_buffer, - sizeof(data->output_buffer[0]) * num_samples, - data->bit_rate / 1000); - if (cmd != DECODE_COMMAND_NONE) - return cmd; - } - - return DECODE_COMMAND_NONE; -} - -/** - * Synthesize the current frame and send it via decoder_data(). - */ -static enum decoder_command -mp3_synth_and_send(struct mp3_data *data) -{ - unsigned i, pcm_length; - enum decoder_command cmd; - - mad_synth_frame(&data->synth, &data->frame); - - if (!data->found_first_frame) { - unsigned int samples_per_frame = data->synth.pcm.length; - data->drop_start_frames = data->drop_start_samples / samples_per_frame; - data->drop_end_frames = data->drop_end_samples / samples_per_frame; - data->drop_start_samples = data->drop_start_samples % samples_per_frame; - data->drop_end_samples = data->drop_end_samples % samples_per_frame; - data->found_first_frame = true; - } - - if (data->drop_start_frames > 0) { - data->drop_start_frames--; - return DECODE_COMMAND_NONE; - } else if ((data->drop_end_frames > 0) && - (data->current_frame == (data->max_frames + 1 - data->drop_end_frames))) { - /* stop decoding, effectively dropping all remaining - frames */ - return DECODE_COMMAND_STOP; - } - - if (!data->decoded_first_frame) { - i = data->drop_start_samples; - data->decoded_first_frame = true; - } else - i = 0; - - pcm_length = data->synth.pcm.length; - if (data->drop_end_samples && - (data->current_frame == data->max_frames - data->drop_end_frames)) { - if (data->drop_end_samples >= pcm_length) - pcm_length = 0; - else - pcm_length -= data->drop_end_samples; - } - - cmd = mp3_send_pcm(data, i, pcm_length); - if (cmd != DECODE_COMMAND_NONE) - return cmd; - - if (data->drop_end_samples && - (data->current_frame == data->max_frames - data->drop_end_frames)) - /* stop decoding, effectively dropping - * all remaining samples */ - return DECODE_COMMAND_STOP; - - return DECODE_COMMAND_NONE; -} - -static bool -mp3_read(struct mp3_data *data) -{ - struct decoder *decoder = data->decoder; - enum mp3_action ret; - enum decoder_command cmd; - - mp3_update_timer_next_frame(data); - - switch (data->mute_frame) { - case MUTEFRAME_SKIP: - data->mute_frame = MUTEFRAME_NONE; - break; - case MUTEFRAME_SEEK: - if (data->elapsed_time >= data->seek_where) - data->mute_frame = MUTEFRAME_NONE; - break; - case MUTEFRAME_NONE: - cmd = mp3_synth_and_send(data); - if (cmd == DECODE_COMMAND_SEEK) { - unsigned long j; - - assert(data->input_stream->seekable); - - j = mp3_time_to_frame(data, - decoder_seek_where(decoder)); - if (j < data->highest_frame) { - if (mp3_seek(data, data->frame_offsets[j])) { - data->current_frame = j; - decoder_command_finished(decoder); - } else - decoder_seek_error(decoder); - } else { - data->seek_where = decoder_seek_where(decoder); - data->mute_frame = MUTEFRAME_SEEK; - decoder_command_finished(decoder); - } - } else if (cmd != DECODE_COMMAND_NONE) - return false; - } - - while (true) { - bool skip = false; - - do { - struct tag *tag = NULL; - - ret = decode_next_frame_header(data, &tag); - - if (tag != NULL) { - decoder_tag(decoder, data->input_stream, tag); - tag_free(tag); - } - } while (ret == DECODE_CONT); - if (ret == DECODE_BREAK) - return false; - else if (ret == DECODE_SKIP) - skip = true; - - if (data->mute_frame == MUTEFRAME_NONE) { - do { - ret = decodeNextFrame(data); - } while (ret == DECODE_CONT); - if (ret == DECODE_BREAK) - return false; - } - - if (!skip && ret == DECODE_OK) - break; - } - - return ret != DECODE_BREAK; -} - -static void -mp3_decode(struct decoder *decoder, struct input_stream *input_stream) -{ - struct mp3_data data; - GError *error = NULL; - struct tag *tag = NULL; - struct audio_format audio_format; - - if (!mp3_open(input_stream, &data, decoder, &tag)) { - if (decoder_get_command(decoder) == DECODE_COMMAND_NONE) - g_warning - ("Input does not appear to be a mp3 bit stream.\n"); - return; - } - - if (!audio_format_init_checked(&audio_format, - data.frame.header.samplerate, - SAMPLE_FORMAT_S24_P32, - MAD_NCHANNELS(&data.frame.header), - &error)) { - g_warning("%s", error->message); - g_error_free(error); - - if (tag != NULL) - tag_free(tag); - mp3_data_finish(&data); - return; - } - - decoder_initialized(decoder, &audio_format, - data.input_stream->seekable, data.total_time); - - if (tag != NULL) { - decoder_tag(decoder, input_stream, tag); - tag_free(tag); - } - - while (mp3_read(&data)) ; - - mp3_data_finish(&data); -} - -static bool -mad_decoder_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - int total_time; - - total_time = mad_decoder_total_file_time(is); - if (total_time < 0) - return false; - - tag_handler_invoke_duration(handler, handler_ctx, total_time); - return true; -} - -static const char *const mp3_suffixes[] = { "mp3", "mp2", NULL }; -static const char *const mp3_mime_types[] = { "audio/mpeg", NULL }; - -const struct decoder_plugin mad_decoder_plugin = { - .name = "mad", - .init = mp3_plugin_init, - .stream_decode = mp3_decode, - .scan_stream = mad_decoder_scan_stream, - .suffixes = mp3_suffixes, - .mime_types = mp3_mime_types -}; diff --git a/src/decoder/mikmod_decoder_plugin.c b/src/decoder/mikmod_decoder_plugin.c deleted file mode 100644 index a8fe818de..000000000 --- a/src/decoder/mikmod_decoder_plugin.c +++ /dev/null @@ -1,239 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "mpd_error.h" -#include "tag_handler.h" - -#include <glib.h> -#include <mikmod.h> -#include <assert.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "mikmod" - -/* this is largely copied from alsaplayer */ - -#define MIKMOD_FRAME_SIZE 4096 - -static BOOL -mikmod_mpd_init(void) -{ - return VC_Init(); -} - -static void -mikmod_mpd_exit(void) -{ - VC_Exit(); -} - -static void -mikmod_mpd_update(void) -{ -} - -static BOOL -mikmod_mpd_is_present(void) -{ - return true; -} - -static char drv_name[] = PACKAGE_NAME; -static char drv_version[] = VERSION; - -#if (LIBMIKMOD_VERSION > 0x030106) -static char drv_alias[] = PACKAGE; -#endif - -static MDRIVER drv_mpd = { - NULL, - drv_name, - drv_version, - 0, - 255, -#if (LIBMIKMOD_VERSION > 0x030106) - drv_alias, -#if (LIBMIKMOD_VERSION >= 0x030200) - NULL, /* CmdLineHelp */ -#endif - NULL, /* CommandLine */ -#endif - mikmod_mpd_is_present, - VC_SampleLoad, - VC_SampleUnload, - VC_SampleSpace, - VC_SampleLength, - mikmod_mpd_init, - mikmod_mpd_exit, - NULL, - VC_SetNumVoices, - VC_PlayStart, - VC_PlayStop, - mikmod_mpd_update, - NULL, - VC_VoiceSetVolume, - VC_VoiceGetVolume, - VC_VoiceSetFrequency, - VC_VoiceGetFrequency, - VC_VoiceSetPanning, - VC_VoiceGetPanning, - VC_VoicePlay, - VC_VoiceStop, - VC_VoiceStopped, - VC_VoiceGetPosition, - VC_VoiceRealVolume -}; - -static unsigned mikmod_sample_rate; - -static bool -mikmod_decoder_init(const struct config_param *param) -{ - static char params[] = ""; - - mikmod_sample_rate = config_get_block_unsigned(param, "sample_rate", - 44100); - if (!audio_valid_sample_rate(mikmod_sample_rate)) - MPD_ERROR("Invalid sample rate in line %d: %u", - param->line, mikmod_sample_rate); - - md_device = 0; - md_reverb = 0; - - MikMod_RegisterDriver(&drv_mpd); - MikMod_RegisterAllLoaders(); - - md_pansep = 64; - md_mixfreq = mikmod_sample_rate; - md_mode = (DMODE_SOFT_MUSIC | DMODE_INTERP | DMODE_STEREO | - DMODE_16BITS); - - if (MikMod_Init(params)) { - g_warning("Could not init MikMod: %s\n", - MikMod_strerror(MikMod_errno)); - return false; - } - - return true; -} - -static void -mikmod_decoder_finish(void) -{ - MikMod_Exit(); -} - -static void -mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs) -{ - char *path2; - MODULE *handle; - struct audio_format audio_format; - int ret; - SBYTE buffer[MIKMOD_FRAME_SIZE]; - enum decoder_command cmd = DECODE_COMMAND_NONE; - - path2 = g_strdup(path_fs); - handle = Player_Load(path2, 128, 0); - g_free(path2); - - if (handle == NULL) { - g_warning("failed to open mod: %s", path_fs); - return; - } - - /* Prevent module from looping forever */ - handle->loop = 0; - - audio_format_init(&audio_format, mikmod_sample_rate, SAMPLE_FORMAT_S16, 2); - assert(audio_format_valid(&audio_format)); - - decoder_initialized(decoder, &audio_format, false, 0); - - Player_Start(handle); - while (cmd == DECODE_COMMAND_NONE && Player_Active()) { - ret = VC_WriteBytes(buffer, sizeof(buffer)); - cmd = decoder_data(decoder, NULL, buffer, ret, 0); - } - - Player_Stop(); - Player_Free(handle); -} - -static bool -mikmod_decoder_scan_file(const char *path_fs, - const struct tag_handler *handler, void *handler_ctx) -{ - char *path2 = g_strdup(path_fs); - MODULE *handle = Player_Load(path2, 128, 0); - - if (handle == NULL) { - g_free(path2); - g_debug("Failed to open file: %s", path_fs); - return false; - - } - - Player_Free(handle); - - char *title = Player_LoadTitle(path2); - g_free(path2); - - if (title != NULL) { - tag_handler_invoke_tag(handler, handler_ctx, - TAG_TITLE, title); -#if (LIBMIKMOD_VERSION >= 0x030200) - MikMod_free(title); -#else - free(title); -#endif - } - - return true; -} - -static const char *const mikmod_decoder_suffixes[] = { - "amf", - "dsm", - "far", - "gdm", - "imf", - "it", - "med", - "mod", - "mtm", - "s3m", - "stm", - "stx", - "ult", - "uni", - "xm", - NULL -}; - -const struct decoder_plugin mikmod_decoder_plugin = { - .name = "mikmod", - .init = mikmod_decoder_init, - .finish = mikmod_decoder_finish, - .file_decode = mikmod_decoder_file_decode, - .scan_file = mikmod_decoder_scan_file, - .suffixes = mikmod_decoder_suffixes, -}; diff --git a/src/decoder/modplug_decoder_plugin.c b/src/decoder/modplug_decoder_plugin.c deleted file mode 100644 index 21ee79e7e..000000000 --- a/src/decoder/modplug_decoder_plugin.c +++ /dev/null @@ -1,194 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "tag_handler.h" - -#include <glib.h> -#include <modplug.h> -#include <assert.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "modplug" - -enum { - MODPLUG_FRAME_SIZE = 4096, - MODPLUG_PREALLOC_BLOCK = 256 * 1024, - MODPLUG_READ_BLOCK = 128 * 1024, - MODPLUG_FILE_LIMIT = 100 * 1024 * 1024, -}; - -static GByteArray *mod_loadfile(struct decoder *decoder, struct input_stream *is) -{ - unsigned char *data; - GByteArray *bdatas; - size_t ret; - - if (is->size == 0) { - g_warning("file is empty"); - return NULL; - } - - if (is->size > MODPLUG_FILE_LIMIT) { - g_warning("file too large"); - return NULL; - } - - //known/unknown size, preallocate array, lets read in chunks - if (is->size > 0) { - bdatas = g_byte_array_sized_new(is->size); - } else { - bdatas = g_byte_array_sized_new(MODPLUG_PREALLOC_BLOCK); - } - - data = g_malloc(MODPLUG_READ_BLOCK); - - while (true) { - ret = decoder_read(decoder, is, data, MODPLUG_READ_BLOCK); - if (ret == 0) { - if (input_stream_lock_eof(is)) - /* end of file */ - break; - - /* I/O error - skip this song */ - g_free(data); - g_byte_array_free(bdatas, true); - return NULL; - } - - if (bdatas->len + ret > MODPLUG_FILE_LIMIT) { - g_warning("stream too large\n"); - g_free(data); - g_byte_array_free(bdatas, TRUE); - return NULL; - } - - g_byte_array_append(bdatas, data, ret); - } - - g_free(data); - - return bdatas; -} - -static void -mod_decode(struct decoder *decoder, struct input_stream *is) -{ - ModPlugFile *f; - ModPlug_Settings settings; - GByteArray *bdatas; - struct audio_format audio_format; - int ret; - char audio_buffer[MODPLUG_FRAME_SIZE]; - enum decoder_command cmd = DECODE_COMMAND_NONE; - - bdatas = mod_loadfile(decoder, is); - - if (!bdatas) { - g_warning("could not load stream\n"); - return; - } - - ModPlug_GetSettings(&settings); - /* alter setting */ - settings.mResamplingMode = MODPLUG_RESAMPLE_FIR; /* RESAMP */ - settings.mChannels = 2; - settings.mBits = 16; - settings.mFrequency = 44100; - /* insert more setting changes here */ - ModPlug_SetSettings(&settings); - - f = ModPlug_Load(bdatas->data, bdatas->len); - g_byte_array_free(bdatas, TRUE); - if (!f) { - g_warning("could not decode stream\n"); - return; - } - - audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); - assert(audio_format_valid(&audio_format)); - - decoder_initialized(decoder, &audio_format, - is->seekable, ModPlug_GetLength(f) / 1000.0); - - do { - ret = ModPlug_Read(f, audio_buffer, MODPLUG_FRAME_SIZE); - if (ret <= 0) - break; - - cmd = decoder_data(decoder, NULL, - audio_buffer, ret, - 0); - - if (cmd == DECODE_COMMAND_SEEK) { - float where = decoder_seek_where(decoder); - - ModPlug_Seek(f, (int)(where * 1000.0)); - - decoder_command_finished(decoder); - } - - } while (cmd != DECODE_COMMAND_STOP); - - ModPlug_Unload(f); -} - -static bool -modplug_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - ModPlugFile *f; - GByteArray *bdatas; - - bdatas = mod_loadfile(NULL, is); - if (!bdatas) - return false; - - f = ModPlug_Load(bdatas->data, bdatas->len); - g_byte_array_free(bdatas, TRUE); - if (f == NULL) - return false; - - tag_handler_invoke_duration(handler, handler_ctx, - ModPlug_GetLength(f) / 1000); - - const char *title = ModPlug_GetName(f); - if (title != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_TITLE, title); - - ModPlug_Unload(f); - - return true; -} - -static const char *const mod_suffixes[] = { - "669", "amf", "ams", "dbm", "dfm", "dsm", "far", "it", - "med", "mdl", "mod", "mtm", "mt2", "okt", "s3m", "stm", - "ult", "umx", "xm", - NULL -}; - -const struct decoder_plugin modplug_decoder_plugin = { - .name = "modplug", - .stream_decode = mod_decode, - .scan_stream = modplug_scan_stream, - .suffixes = mod_suffixes, -}; diff --git a/src/decoder/mp4ff_decoder_plugin.c b/src/decoder/mp4ff_decoder_plugin.c deleted file mode 100644 index ca78a22d0..000000000 --- a/src/decoder/mp4ff_decoder_plugin.c +++ /dev/null @@ -1,448 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "tag_table.h" -#include "tag_handler.h" - -#include <glib.h> - -#include <mp4ff.h> -#include <faad.h> - -#include <assert.h> -#include <stdlib.h> -#include <unistd.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "mp4ff" - -/* all code here is either based on or copied from FAAD2's frontend code */ - -struct mp4ff_input_stream { - mp4ff_callback_t callback; - - struct decoder *decoder; - struct input_stream *input_stream; -}; - -static int -mp4_get_aac_track(mp4ff_t * infile, faacDecHandle decoder, - uint32_t *sample_rate, unsigned char *channels_r) -{ -#ifdef HAVE_FAAD_LONG - /* neaacdec.h declares all arguments as "unsigned long", but - internally expects uint32_t pointers. To avoid gcc - warnings, use this workaround. */ - unsigned long *sample_rate_r = (unsigned long*)sample_rate; -#else - uint32_t *sample_rate_r = sample_rate; -#endif - int i, rc; - int num_tracks = mp4ff_total_tracks(infile); - - for (i = 0; i < num_tracks; i++) { - unsigned char *buff = NULL; - unsigned int buff_size = 0; - - if (mp4ff_get_track_type(infile, i) != 1) - /* not an audio track */ - continue; - - if (decoder == NULL) - /* have don't have a decoder to initialize - - we're done now, because we found an audio - track */ - return i; - - mp4ff_get_decoder_config(infile, i, &buff, &buff_size); - if (buff == NULL) - continue; - - rc = faacDecInit2(decoder, buff, buff_size, - sample_rate_r, channels_r); - free(buff); - - if (rc >= 0) - /* found a valid AAC track */ - return i; - } - - /* can't decode this */ - return -1; -} - -static uint32_t -mp4_read(void *user_data, void *buffer, uint32_t length) -{ - struct mp4ff_input_stream *mis = user_data; - - if (length == 0) - /* libmp4ff is known to attempt to read 0 bytes - make - this a special case, because the input_stream API - would not allow this */ - return 0; - - return decoder_read(mis->decoder, mis->input_stream, buffer, length); -} - -static uint32_t -mp4_seek(void *user_data, uint64_t position) -{ - struct mp4ff_input_stream *mis = user_data; - - return input_stream_lock_seek(mis->input_stream, position, SEEK_SET, - NULL) - ? 0 : -1; -} - -static const mp4ff_callback_t mpd_mp4ff_callback = { - .read = mp4_read, - .seek = mp4_seek, -}; - -static mp4ff_t * -mp4ff_input_stream_open(struct mp4ff_input_stream *mis, - struct decoder *decoder, - struct input_stream *input_stream) -{ - mis->callback = mpd_mp4ff_callback; - mis->callback.user_data = mis; - mis->decoder = decoder; - mis->input_stream = input_stream; - - return mp4ff_open_read(&mis->callback); -} - -static faacDecHandle -mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format) -{ - faacDecHandle decoder; - faacDecConfigurationPtr config; - int track; - uint32_t sample_rate; - unsigned char channels; - GError *error = NULL; - - decoder = faacDecOpen(); - - config = faacDecGetCurrentConfiguration(decoder); - config->outputFormat = FAAD_FMT_16BIT; -#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX - config->downMatrix = 1; -#endif -#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR - config->dontUpSampleImplicitSBR = 0; -#endif - faacDecSetConfiguration(decoder, config); - - track = mp4_get_aac_track(mp4fh, decoder, &sample_rate, &channels); - if (track < 0) { - g_warning("No AAC track found"); - faacDecClose(decoder); - return NULL; - } - - if (!audio_format_init_checked(audio_format, sample_rate, - SAMPLE_FORMAT_S16, channels, - &error)) { - g_warning("%s", error->message); - g_error_free(error); - faacDecClose(decoder); - return NULL; - } - - *track_r = track; - - return decoder; -} - -static void -mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) -{ - struct mp4ff_input_stream mis; - mp4ff_t *mp4fh; - int32_t track; - float file_time, total_time; - int32_t scale; - faacDecHandle decoder; - struct audio_format audio_format; - faacDecFrameInfo frame_info; - unsigned char *mp4_buffer; - unsigned int mp4_buffer_size; - long sample_id; - long num_samples; - long dur; - unsigned int sample_count; - char *sample_buffer; - size_t sample_buffer_length; - unsigned int initial = 1; - float *seek_table; - long seek_table_end = -1; - bool seek_position_found = false; - long offset; - uint16_t bit_rate = 0; - bool seeking = false; - double seek_where = 0; - enum decoder_command cmd = DECODE_COMMAND_NONE; - - mp4fh = mp4ff_input_stream_open(&mis, mpd_decoder, input_stream); - if (!mp4fh) { - g_warning("Input does not appear to be a mp4 stream.\n"); - return; - } - - decoder = mp4_faad_new(mp4fh, &track, &audio_format); - if (decoder == NULL) { - mp4ff_close(mp4fh); - return; - } - - file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track); - scale = mp4ff_time_scale(mp4fh, track); - - if (scale < 0) { - g_warning("Error getting audio format of mp4 AAC track.\n"); - faacDecClose(decoder); - mp4ff_close(mp4fh); - return; - } - total_time = ((float)file_time) / scale; - - num_samples = mp4ff_num_samples(mp4fh, track); - if (num_samples > (long)(G_MAXINT / sizeof(float))) { - g_warning("Integer overflow.\n"); - faacDecClose(decoder); - mp4ff_close(mp4fh); - return; - } - - file_time = 0.0; - - seek_table = input_stream->seekable - ? g_malloc(sizeof(float) * num_samples) - : NULL; - - decoder_initialized(mpd_decoder, &audio_format, - input_stream->seekable, - total_time); - - for (sample_id = 0; - sample_id < num_samples && cmd != DECODE_COMMAND_STOP; - sample_id++) { - if (cmd == DECODE_COMMAND_SEEK) { - assert(seek_table != NULL); - - seeking = true; - seek_where = decoder_seek_where(mpd_decoder); - } - - if (seeking && seek_table_end > 1 && - seek_table[seek_table_end] >= seek_where) { - int i = 2; - - assert(seek_table != NULL); - - while (seek_table[i] < seek_where) - i++; - sample_id = i - 1; - file_time = seek_table[sample_id]; - } - - dur = mp4ff_get_sample_duration(mp4fh, track, sample_id); - offset = mp4ff_get_sample_offset(mp4fh, track, sample_id); - - if (seek_table != NULL && sample_id > seek_table_end) { - seek_table[sample_id] = file_time; - seek_table_end = sample_id; - } - - if (sample_id == 0) - dur = 0; - if (offset > dur) - dur = 0; - else - dur -= offset; - file_time += ((float)dur) / scale; - - if (seeking && file_time >= seek_where) - seek_position_found = true; - - if (seeking && seek_position_found) { - seek_position_found = false; - seeking = 0; - decoder_command_finished(mpd_decoder); - } - - if (seeking) - continue; - - if (mp4ff_read_sample(mp4fh, track, sample_id, &mp4_buffer, - &mp4_buffer_size) == 0) - break; - -#ifdef HAVE_FAAD_BUFLEN_FUNCS - sample_buffer = faacDecDecode(decoder, &frame_info, mp4_buffer, - mp4_buffer_size); -#else - sample_buffer = faacDecDecode(decoder, &frame_info, mp4_buffer); -#endif - - free(mp4_buffer); - - if (frame_info.error > 0) { - g_warning("faad2 error: %s\n", - faacDecGetErrorMessage(frame_info.error)); - break; - } - - if (frame_info.channels != audio_format.channels) { - g_warning("channel count changed from %u to %u", - audio_format.channels, frame_info.channels); - break; - } - -#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE - if (frame_info.samplerate != audio_format.sample_rate) { - g_warning("sample rate changed from %u to %lu", - audio_format.sample_rate, - (unsigned long)frame_info.samplerate); - break; - } -#endif - - if (audio_format.channels * (unsigned long)(dur + offset) > frame_info.samples) { - dur = frame_info.samples / audio_format.channels; - offset = 0; - } - - sample_count = (unsigned long)(dur * audio_format.channels); - - if (sample_count > 0) { - initial = 0; - bit_rate = frame_info.bytesconsumed * 8.0 * - frame_info.channels * scale / - frame_info.samples / 1000 + 0.5; - } - - sample_buffer_length = sample_count * 2; - - sample_buffer += offset * audio_format.channels * 2; - - cmd = decoder_data(mpd_decoder, input_stream, - sample_buffer, sample_buffer_length, - bit_rate); - } - - g_free(seek_table); - faacDecClose(decoder); - mp4ff_close(mp4fh); -} - -static const struct tag_table mp4ff_tags[] = { - { "album artist", TAG_ALBUM_ARTIST }, - { "writer", TAG_COMPOSER }, - { "band", TAG_PERFORMER }, - { NULL, TAG_NUM_OF_ITEM_TYPES } -}; - -static enum tag_type -mp4ff_tag_name_parse(const char *name) -{ - enum tag_type type = tag_table_lookup_i(mp4ff_tags, name); - if (type == TAG_NUM_OF_ITEM_TYPES) - type = tag_name_parse_i(name); - - if (g_ascii_strcasecmp(name, "albumartist") == 0 || - g_ascii_strcasecmp(name, "album_artist") == 0) - return TAG_ALBUM_ARTIST; - - return type; -} - -static bool -mp4ff_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - struct mp4ff_input_stream mis; - int32_t track; - int32_t file_time; - int32_t scale; - int i; - - mp4ff_t *mp4fh = mp4ff_input_stream_open(&mis, NULL, is); - if (mp4fh == NULL) - return false; - - track = mp4_get_aac_track(mp4fh, NULL, NULL, NULL); - if (track < 0) { - mp4ff_close(mp4fh); - return false; - } - - file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track); - scale = mp4ff_time_scale(mp4fh, track); - if (scale < 0) { - mp4ff_close(mp4fh); - return false; - } - - tag_handler_invoke_duration(handler, handler_ctx, - ((float)file_time) / scale + 0.5); - - for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) { - char *item; - char *value; - - mp4ff_meta_get_by_index(mp4fh, i, &item, &value); - - tag_handler_invoke_pair(handler, handler_ctx, item, value); - - enum tag_type type = mp4ff_tag_name_parse(item); - if (type != TAG_NUM_OF_ITEM_TYPES) - tag_handler_invoke_tag(handler, handler_ctx, - type, value); - - free(item); - free(value); - } - - mp4ff_close(mp4fh); - - return true; -} - -static const char *const mp4_suffixes[] = { - "m4a", - "m4b", - "mp4", - NULL -}; - -static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL }; - -const struct decoder_plugin mp4ff_decoder_plugin = { - .name = "mp4ff", - .stream_decode = mp4_decode, - .scan_stream = mp4ff_scan_stream, - .suffixes = mp4_suffixes, - .mime_types = mp4_mime_types, -}; diff --git a/src/decoder/mpcdec_decoder_plugin.c b/src/decoder/mpcdec_decoder_plugin.c deleted file mode 100644 index d4768b35b..000000000 --- a/src/decoder/mpcdec_decoder_plugin.c +++ /dev/null @@ -1,347 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "tag_handler.h" - -#ifdef MPC_IS_OLD_API -#include <mpcdec/mpcdec.h> -#else -#include <mpc/mpcdec.h> -#include <math.h> -#endif - -#include <glib.h> -#include <assert.h> -#include <unistd.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "mpcdec" - -struct mpc_decoder_data { - struct input_stream *is; - struct decoder *decoder; -}; - -#ifdef MPC_IS_OLD_API -#define cb_first_arg void *vdata -#define cb_data vdata -#else -#define cb_first_arg mpc_reader *reader -#define cb_data reader->data -#endif - -static mpc_int32_t -mpc_read_cb(cb_first_arg, void *ptr, mpc_int32_t size) -{ - struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data; - - return decoder_read(data->decoder, data->is, ptr, size); -} - -static mpc_bool_t -mpc_seek_cb(cb_first_arg, mpc_int32_t offset) -{ - struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data; - - return input_stream_lock_seek(data->is, offset, SEEK_SET, NULL); -} - -static mpc_int32_t -mpc_tell_cb(cb_first_arg) -{ - struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data; - - return (long)(data->is->offset); -} - -static mpc_bool_t -mpc_canseek_cb(cb_first_arg) -{ - struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data; - - return data->is->seekable; -} - -static mpc_int32_t -mpc_getsize_cb(cb_first_arg) -{ - struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data; - - return data->is->size; -} - -/* this _looks_ performance-critical, don't de-inline -- eric */ -static inline int32_t -mpc_to_mpd_sample(MPC_SAMPLE_FORMAT sample) -{ - /* only doing 16-bit audio for now */ - int32_t val; - - enum { - bits = 24, - }; - - const int clip_min = -1 << (bits - 1); - const int clip_max = (1 << (bits - 1)) - 1; - -#ifdef MPC_FIXED_POINT - const int shift = bits - MPC_FIXED_POINT_SCALE_SHIFT; - - if (shift < 0) - val = sample >> -shift; - else - val = sample << shift; -#else - const int float_scale = 1 << (bits - 1); - - val = sample * float_scale; -#endif - - if (val < clip_min) - val = clip_min; - else if (val > clip_max) - val = clip_max; - - return val; -} - -static void -mpc_to_mpd_buffer(int32_t *dest, const MPC_SAMPLE_FORMAT *src, - unsigned num_samples) -{ - while (num_samples-- > 0) - *dest++ = mpc_to_mpd_sample(*src++); -} - -static void -mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) -{ -#ifdef MPC_IS_OLD_API - mpc_decoder decoder; -#else - mpc_demux *demux; - mpc_frame_info frame; - mpc_status status; -#endif - mpc_reader reader; - mpc_streaminfo info; - GError *error = NULL; - struct audio_format audio_format; - - struct mpc_decoder_data data; - - MPC_SAMPLE_FORMAT sample_buffer[MPC_DECODER_BUFFER_LENGTH]; - - mpc_uint32_t ret; - int32_t chunk[G_N_ELEMENTS(sample_buffer)]; - long bit_rate = 0; - mpc_uint32_t vbr_update_bits; - enum decoder_command cmd = DECODE_COMMAND_NONE; - - data.is = is; - data.decoder = mpd_decoder; - - reader.read = mpc_read_cb; - reader.seek = mpc_seek_cb; - reader.tell = mpc_tell_cb; - reader.get_size = mpc_getsize_cb; - reader.canseek = mpc_canseek_cb; - reader.data = &data; - -#ifdef MPC_IS_OLD_API - mpc_streaminfo_init(&info); - - if ((ret = mpc_streaminfo_read(&info, &reader)) != ERROR_CODE_OK) { - if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) - g_warning("Not a valid musepack stream\n"); - return; - } - - mpc_decoder_setup(&decoder, &reader); - - if (!mpc_decoder_initialize(&decoder, &info)) { - if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) - g_warning("Not a valid musepack stream\n"); - return; - } -#else - demux = mpc_demux_init(&reader); - if (demux == NULL) { - if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP) - g_warning("Not a valid musepack stream"); - return; - } - - mpc_demux_get_info(demux, &info); -#endif - - if (!audio_format_init_checked(&audio_format, info.sample_freq, - SAMPLE_FORMAT_S24_P32, - info.channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); -#ifndef MPC_IS_OLD_API - mpc_demux_exit(demux); -#endif - return; - } - - struct replay_gain_info replay_gain_info; - replay_gain_info_init(&replay_gain_info); -#ifdef MPC_IS_OLD_API - replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = info.gain_album * 0.01; - replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = info.peak_album / 32767.0; - replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = info.gain_title * 0.01; - replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = info.peak_title / 32767.0; -#else - replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = MPC_OLD_GAIN_REF - (info.gain_album / 256.); - replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = pow(10, info.peak_album / 256. / 20) / 32767; - replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = MPC_OLD_GAIN_REF - (info.gain_title / 256.); - replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = pow(10, info.peak_title / 256. / 20) / 32767; -#endif - - decoder_replay_gain(mpd_decoder, &replay_gain_info); - - decoder_initialized(mpd_decoder, &audio_format, - is->seekable, - mpc_streaminfo_get_length(&info)); - - do { - if (cmd == DECODE_COMMAND_SEEK) { - mpc_int64_t where = decoder_seek_where(mpd_decoder) * - audio_format.sample_rate; - bool success; - -#ifdef MPC_IS_OLD_API - success = mpc_decoder_seek_sample(&decoder, where); -#else - success = mpc_demux_seek_sample(demux, where) - == MPC_STATUS_OK; -#endif - if (success) - decoder_command_finished(mpd_decoder); - else - decoder_seek_error(mpd_decoder); - } - - vbr_update_bits = 0; - -#ifdef MPC_IS_OLD_API - mpc_uint32_t vbr_update_acc = 0; - - ret = mpc_decoder_decode(&decoder, sample_buffer, - &vbr_update_acc, &vbr_update_bits); - if (ret == 0 || ret == (mpc_uint32_t)-1) - break; -#else - frame.buffer = (MPC_SAMPLE_FORMAT *)sample_buffer; - status = mpc_demux_decode(demux, &frame); - if (status != MPC_STATUS_OK) { - g_warning("Failed to decode sample"); - break; - } - - if (frame.bits == -1) - break; - - ret = frame.samples; -#endif - - ret *= info.channels; - - mpc_to_mpd_buffer(chunk, sample_buffer, ret); - - bit_rate = vbr_update_bits * audio_format.sample_rate - / 1152 / 1000; - - cmd = decoder_data(mpd_decoder, is, - chunk, ret * sizeof(chunk[0]), - bit_rate); - } while (cmd != DECODE_COMMAND_STOP); - -#ifndef MPC_IS_OLD_API - mpc_demux_exit(demux); -#endif -} - -static float -mpcdec_get_file_duration(struct input_stream *is) -{ - float total_time = -1; - - mpc_reader reader; -#ifndef MPC_IS_OLD_API - mpc_demux *demux; -#endif - mpc_streaminfo info; - struct mpc_decoder_data data; - - data.is = is; - data.decoder = NULL; - - reader.read = mpc_read_cb; - reader.seek = mpc_seek_cb; - reader.tell = mpc_tell_cb; - reader.get_size = mpc_getsize_cb; - reader.canseek = mpc_canseek_cb; - reader.data = &data; - -#ifdef MPC_IS_OLD_API - mpc_streaminfo_init(&info); - - if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) - return -1; -#else - demux = mpc_demux_init(&reader); - if (demux == NULL) - return -1; - - mpc_demux_get_info(demux, &info); - mpc_demux_exit(demux); -#endif - - total_time = mpc_streaminfo_get_length(&info); - - return total_time; -} - -static bool -mpcdec_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - float total_time = mpcdec_get_file_duration(is); - - if (total_time < 0) - return false; - - tag_handler_invoke_duration(handler, handler_ctx, total_time); - return true; -} - -static const char *const mpcdec_suffixes[] = { "mpc", NULL }; - -const struct decoder_plugin mpcdec_decoder_plugin = { - .name = "mpcdec", - .stream_decode = mpcdec_decode, - .scan_stream = mpcdec_scan_stream, - .suffixes = mpcdec_suffixes, -}; diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c deleted file mode 100644 index 657a9c889..000000000 --- a/src/decoder/mpg123_decoder_plugin.c +++ /dev/null @@ -1,245 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" /* must be first for large file support */ -#include "decoder_api.h" -#include "audio_check.h" -#include "tag_handler.h" - -#include <glib.h> - -#include <mpg123.h> -#include <stdio.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "mpg123" - -static bool -mpd_mpg123_init(G_GNUC_UNUSED const struct config_param *param) -{ - mpg123_init(); - - return true; -} - -static void -mpd_mpg123_finish(void) -{ - mpg123_exit(); -} - -/** - * Opens a file with an existing #mpg123_handle. - * - * @param handle a handle which was created before; on error, this - * function will not free it - * @param audio_format this parameter is filled after successful - * return - * @return true on success - */ -static bool -mpd_mpg123_open(mpg123_handle *handle, const char *path_fs, - struct audio_format *audio_format) -{ - GError *gerror = NULL; - char *path_dup; - int error; - int channels, encoding; - long rate; - - /* mpg123_open() wants a writable string :-( */ - path_dup = g_strdup(path_fs); - - error = mpg123_open(handle, path_dup); - g_free(path_dup); - if (error != MPG123_OK) { - g_warning("libmpg123 failed to open %s: %s", - path_fs, mpg123_plain_strerror(error)); - return false; - } - - /* obtain the audio format */ - - error = mpg123_getformat(handle, &rate, &channels, &encoding); - if (error != MPG123_OK) { - g_warning("mpg123_getformat() failed: %s", - mpg123_plain_strerror(error)); - return false; - } - - if (encoding != MPG123_ENC_SIGNED_16) { - /* other formats not yet implemented */ - g_warning("expected MPG123_ENC_SIGNED_16, got %d", encoding); - return false; - } - - if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16, - channels, &gerror)) { - g_warning("%s", gerror->message); - g_error_free(gerror); - return false; - } - - return true; -} - -static void -mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs) -{ - struct audio_format audio_format; - mpg123_handle *handle; - int error; - off_t num_samples; - enum decoder_command cmd; - struct mpg123_frameinfo info; - - /* open the file */ - - handle = mpg123_new(NULL, &error); - if (handle == NULL) { - g_warning("mpg123_new() failed: %s", - mpg123_plain_strerror(error)); - return; - } - - if (!mpd_mpg123_open(handle, path_fs, &audio_format)) { - mpg123_delete(handle); - return; - } - - num_samples = mpg123_length(handle); - - /* tell MPD core we're ready */ - - decoder_initialized(decoder, &audio_format, true, - (float)num_samples / - (float)audio_format.sample_rate); - - if (mpg123_info(handle, &info) != MPG123_OK) { - info.vbr = MPG123_CBR; - info.bitrate = 0; - } - - switch (info.vbr) { - case MPG123_ABR: - info.bitrate = info.abr_rate; - break; - case MPG123_CBR: - break; - default: - info.bitrate = 0; - } - - /* the decoder main loop */ - - do { - unsigned char buffer[8192]; - size_t nbytes; - - /* decode */ - - error = mpg123_read(handle, buffer, sizeof(buffer), &nbytes); - if (error != MPG123_OK) { - if (error != MPG123_DONE) - g_warning("mpg123_read() failed: %s", - mpg123_plain_strerror(error)); - break; - } - - /* update bitrate for ABR/VBR */ - if (info.vbr != MPG123_CBR) { - /* FIXME: maybe skip, as too expensive? */ - /* FIXME: maybe, (info.vbr == MPG123_VBR) ? */ - if (mpg123_info (handle, &info) != MPG123_OK) - info.bitrate = 0; - } - - /* send to MPD */ - - cmd = decoder_data(decoder, NULL, buffer, nbytes, info.bitrate); - - if (cmd == DECODE_COMMAND_SEEK) { - off_t c = decoder_seek_where(decoder)*audio_format.sample_rate; - c = mpg123_seek(handle, c, SEEK_SET); - if (c < 0) - decoder_seek_error(decoder); - else { - decoder_command_finished(decoder); - decoder_timestamp(decoder, c/(double)audio_format.sample_rate); - } - - cmd = DECODE_COMMAND_NONE; - } - } while (cmd == DECODE_COMMAND_NONE); - - /* cleanup */ - - mpg123_delete(handle); -} - -static bool -mpd_mpg123_scan_file(const char *path_fs, - const struct tag_handler *handler, void *handler_ctx) -{ - struct audio_format audio_format; - mpg123_handle *handle; - int error; - off_t num_samples; - - handle = mpg123_new(NULL, &error); - if (handle == NULL) { - g_warning("mpg123_new() failed: %s", - mpg123_plain_strerror(error)); - return false; - } - - if (!mpd_mpg123_open(handle, path_fs, &audio_format)) { - mpg123_delete(handle); - return false; - } - - num_samples = mpg123_length(handle); - if (num_samples <= 0) { - mpg123_delete(handle); - return false; - } - - /* ID3 tag support not yet implemented */ - - mpg123_delete(handle); - - tag_handler_invoke_duration(handler, handler_ctx, - num_samples / audio_format.sample_rate); - return true; -} - -static const char *const mpg123_suffixes[] = { - "mp3", - NULL -}; - -const struct decoder_plugin mpg123_decoder_plugin = { - .name = "mpg123", - .init = mpd_mpg123_init, - .finish = mpd_mpg123_finish, - .file_decode = mpd_mpg123_file_decode, - /* streaming not yet implemented */ - .scan_file = mpd_mpg123_scan_file, - .suffixes = mpg123_suffixes, -}; diff --git a/src/decoder/pcm_decoder_plugin.c b/src/decoder/pcm_decoder_plugin.c deleted file mode 100644 index fc7dffc05..000000000 --- a/src/decoder/pcm_decoder_plugin.c +++ /dev/null @@ -1,105 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder/pcm_decoder_plugin.h" -#include "decoder_api.h" -#include "util/byte_reverse.h" - -#include <glib.h> -#include <unistd.h> -#include <stdio.h> /* for SEEK_SET */ - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "pcm" - -static void -pcm_stream_decode(struct decoder *decoder, struct input_stream *is) -{ - static const struct audio_format audio_format = { - .sample_rate = 44100, - .format = SAMPLE_FORMAT_S16, - .channels = 2, - }; - - const bool reverse_endian = is->mime != NULL && - strcmp(is->mime, "audio/x-mpd-cdda-pcm-reverse") == 0; - - GError *error = NULL; - enum decoder_command cmd; - - double time_to_size = audio_format_time_to_size(&audio_format); - - float total_time = -1; - if (is->size >= 0) - total_time = is->size / time_to_size; - - decoder_initialized(decoder, &audio_format, is->seekable, total_time); - - do { - char buffer[4096]; - - size_t nbytes = decoder_read(decoder, is, - buffer, sizeof(buffer)); - - if (nbytes == 0 && input_stream_lock_eof(is)) - break; - - if (reverse_endian) - /* make sure we deliver samples in host byte order */ - reverse_bytes_16((uint16_t *)buffer, - (uint16_t *)buffer, - (uint16_t *)(buffer + nbytes)); - - cmd = nbytes > 0 - ? decoder_data(decoder, is, - buffer, nbytes, 0) - : decoder_get_command(decoder); - if (cmd == DECODE_COMMAND_SEEK) { - goffset offset = (goffset)(time_to_size * - decoder_seek_where(decoder)); - if (input_stream_lock_seek(is, offset, SEEK_SET, - &error)) { - decoder_command_finished(decoder); - } else { - g_warning("seeking failed: %s", error->message); - g_error_free(error); - decoder_seek_error(decoder); - } - - cmd = DECODE_COMMAND_NONE; - } - } while (cmd == DECODE_COMMAND_NONE); -} - -static const char *const pcm_mime_types[] = { - /* for streams obtained by the cdio_paranoia input plugin */ - "audio/x-mpd-cdda-pcm", - - /* same as above, but with reverse byte order */ - "audio/x-mpd-cdda-pcm-reverse", - - NULL -}; - -const struct decoder_plugin pcm_decoder_plugin = { - .name = "pcm", - .stream_decode = pcm_stream_decode, - .mime_types = pcm_mime_types, -}; diff --git a/src/decoder/pcm_decoder_plugin.h b/src/decoder/pcm_decoder_plugin.h deleted file mode 100644 index 11df80155..000000000 --- a/src/decoder/pcm_decoder_plugin.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/** \file - * - * Not really a decoder; this plugin forwards its input data "as-is". - * - * It was written only to support the "cdio_paranoia" input plugin, - * which does not need a decoder. - */ - -#ifndef MPD_DECODER_PCM_H -#define MPD_DECODER_PCM_H - -extern const struct decoder_plugin pcm_decoder_plugin; - -#endif diff --git a/src/decoder/sidplay_decoder_plugin.cxx b/src/decoder/sidplay_decoder_plugin.cxx index 5d162f179..fed0476ec 100644 --- a/src/decoder/sidplay_decoder_plugin.cxx +++ b/src/decoder/sidplay_decoder_plugin.cxx @@ -18,14 +18,12 @@ */ #include "config.h" - -extern "C" { -#include "../decoder_api.h" -#include "tag_handler.h" -} +#include "../DecoderAPI.hxx" +#include "tag/TagHandler.hxx" #include <errno.h> #include <stdlib.h> +#include <string.h> #include <glib.h> #include <sidplay/sidplay2.h> @@ -82,29 +80,27 @@ sidplay_load_songlength_db(const char *path) } static bool -sidplay_init(const struct config_param *param) +sidplay_init(const config_param ¶m) { /* read the songlengths database file */ - songlength_file=config_get_block_string(param, - "songlength_database", NULL); + songlength_file = param.GetBlockValue("songlength_database"); if (songlength_file != NULL) songlength_database = sidplay_load_songlength_db(songlength_file); - default_songlength=config_get_block_unsigned(param, - "default_songlength", 0); + default_songlength = param.GetBlockValue("default_songlength", 0u); - all_files_are_containers=config_get_block_bool(param, - "all_files_are_containers", true); + all_files_are_containers = + param.GetBlockValue("all_files_are_containers", true); path_with_subtune=g_pattern_spec_new( "*/" SUBTUNE_PREFIX "???.sid"); - filter_setting=config_get_block_bool(param, "filter", true); + filter_setting = param.GetBlockValue("filter", true); return true; } -void +static void sidplay_finish() { g_pattern_spec_free(path_with_subtune); @@ -136,7 +132,7 @@ get_container_name(const char *path_fs) * returns tune number from file.sid/tune_xxx.sid style path or 1 if * no subtune is appended */ -static int +static unsigned get_song_num(const char *path_fs) { if(g_pattern_match(path_with_subtune, @@ -172,7 +168,7 @@ get_song_length(const char *path_fs) char md5sum[SIDTUNE_MD5_LENGTH+1]; tune.createMD5(md5sum); - int song_num=get_song_num(path_fs); + const unsigned song_num = get_song_num(path_fs); gsize num_items; gchar **values=g_key_file_get_string_list(songlength_database, @@ -284,18 +280,17 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) /* initialize the MPD decoder */ - struct audio_format audio_format; - audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, channels); - assert(audio_format_valid(&audio_format)); + const AudioFormat audio_format(48000, SampleFormat::S16, channels); + assert(audio_format.IsValid()); - decoder_initialized(decoder, &audio_format, true, (float)song_len); + decoder_initialized(decoder, audio_format, true, (float)song_len); /* .. and play */ const unsigned timebase = player.timebase(); song_len *= timebase; - enum decoder_command cmd; + DecoderCommand cmd; do { char buffer[4096]; size_t nbytes; @@ -308,7 +303,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) cmd = decoder_data(decoder, NULL, buffer, nbytes, 0); - if(cmd==DECODE_COMMAND_SEEK) { + if (cmd == DecoderCommand::SEEK) { unsigned data_time = player.time(); unsigned target_time = (unsigned) (decoder_seek_where(decoder) * timebase); @@ -330,10 +325,10 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) decoder_command_finished(decoder); } - if (song_len > 0 && player.time() >= song_len) + if (song_len > 0 && player.time() >= (unsigned)song_len) break; - } while (cmd != DECODE_COMMAND_STOP); + } while (cmd != DecoderCommand::STOP); } static bool diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c deleted file mode 100644 index 8dd98236f..000000000 --- a/src/decoder/sndfile_decoder_plugin.c +++ /dev/null @@ -1,255 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "tag_handler.h" - -#include <sndfile.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "sndfile" - -static sf_count_t -sndfile_vio_get_filelen(void *user_data) -{ - const struct input_stream *is = user_data; - - return is->size; -} - -static sf_count_t -sndfile_vio_seek(sf_count_t offset, int whence, void *user_data) -{ - struct input_stream *is = user_data; - bool success; - - success = input_stream_lock_seek(is, offset, whence, NULL); - if (!success) - return -1; - - return is->offset; -} - -static sf_count_t -sndfile_vio_read(void *ptr, sf_count_t count, void *user_data) -{ - struct input_stream *is = user_data; - GError *error = NULL; - size_t nbytes; - - nbytes = input_stream_lock_read(is, ptr, count, &error); - if (nbytes == 0 && error != NULL) { - g_warning("%s", error->message); - g_error_free(error); - return -1; - } - - return nbytes; -} - -static sf_count_t -sndfile_vio_write(G_GNUC_UNUSED const void *ptr, - G_GNUC_UNUSED sf_count_t count, - G_GNUC_UNUSED void *user_data) -{ - /* no writing! */ - return -1; -} - -static sf_count_t -sndfile_vio_tell(void *user_data) -{ - const struct input_stream *is = user_data; - - return is->offset; -} - -/** - * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a - * libsndfile stream. - */ -static SF_VIRTUAL_IO vio = { - .get_filelen = sndfile_vio_get_filelen, - .seek = sndfile_vio_seek, - .read = sndfile_vio_read, - .write = sndfile_vio_write, - .tell = sndfile_vio_tell, -}; - -/** - * Converts a frame number to a timestamp (in seconds). - */ -static float -frame_to_time(sf_count_t frame, const struct audio_format *audio_format) -{ - return (float)frame / (float)audio_format->sample_rate; -} - -/** - * Converts a timestamp (in seconds) to a frame number. - */ -static sf_count_t -time_to_frame(float t, const struct audio_format *audio_format) -{ - return (sf_count_t)(t * audio_format->sample_rate); -} - -static void -sndfile_stream_decode(struct decoder *decoder, struct input_stream *is) -{ - GError *error = NULL; - SNDFILE *sf; - SF_INFO info; - struct audio_format audio_format; - size_t frame_size; - sf_count_t read_frames, num_frames; - int buffer[4096]; - enum decoder_command cmd; - - info.format = 0; - - sf = sf_open_virtual(&vio, SFM_READ, &info, is); - if (sf == NULL) { - g_warning("sf_open_virtual() failed"); - return; - } - - /* for now, always read 32 bit samples. Later, we could lower - MPD's CPU usage by reading 16 bit samples with - sf_readf_short() on low-quality source files. */ - if (!audio_format_init_checked(&audio_format, info.samplerate, - SAMPLE_FORMAT_S32, - info.channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); - return; - } - - decoder_initialized(decoder, &audio_format, info.seekable, - frame_to_time(info.frames, &audio_format)); - - frame_size = audio_format_frame_size(&audio_format); - read_frames = sizeof(buffer) / frame_size; - - do { - num_frames = sf_readf_int(sf, buffer, read_frames); - if (num_frames <= 0) - break; - - cmd = decoder_data(decoder, is, - buffer, num_frames * frame_size, - 0); - if (cmd == DECODE_COMMAND_SEEK) { - sf_count_t c = - time_to_frame(decoder_seek_where(decoder), - &audio_format); - c = sf_seek(sf, c, SEEK_SET); - if (c < 0) - decoder_seek_error(decoder); - else - decoder_command_finished(decoder); - cmd = DECODE_COMMAND_NONE; - } - } while (cmd == DECODE_COMMAND_NONE); - - sf_close(sf); -} - -static bool -sndfile_scan_file(const char *path_fs, - const struct tag_handler *handler, void *handler_ctx) -{ - SNDFILE *sf; - SF_INFO info; - const char *p; - - info.format = 0; - - sf = sf_open(path_fs, SFM_READ, &info); - if (sf == NULL) - return false; - - if (!audio_valid_sample_rate(info.samplerate)) { - sf_close(sf); - g_warning("Invalid sample rate in %s\n", path_fs); - return false; - } - - tag_handler_invoke_duration(handler, handler_ctx, - info.frames / info.samplerate); - - p = sf_get_string(sf, SF_STR_TITLE); - if (p != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_TITLE, p); - - p = sf_get_string(sf, SF_STR_ARTIST); - if (p != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_ARTIST, p); - - p = sf_get_string(sf, SF_STR_DATE); - if (p != NULL) - tag_handler_invoke_tag(handler, handler_ctx, - TAG_DATE, p); - - sf_close(sf); - - return true; -} - -static const char *const sndfile_suffixes[] = { - "wav", "aiff", "aif", /* Microsoft / SGI / Apple */ - "au", "snd", /* Sun / DEC / NeXT */ - "paf", /* Paris Audio File */ - "iff", "svx", /* Commodore Amiga IFF / SVX */ - "sf", /* IRCAM */ - "voc", /* Creative */ - "w64", /* Soundforge */ - "pvf", /* Portable Voice Format */ - "xi", /* Fasttracker */ - "htk", /* HMM Tool Kit */ - "caf", /* Apple */ - "sd2", /* Sound Designer II */ - - /* libsndfile also supports FLAC and Ogg Vorbis, but only by - linking with libFLAC and libvorbis - we can do better, we - have native plugins for these libraries */ - - NULL -}; - -static const char *const sndfile_mime_types[] = { - "audio/x-wav", - "audio/x-aiff", - - /* what are the MIME types of the other supported formats? */ - - NULL -}; - -const struct decoder_plugin sndfile_decoder_plugin = { - .name = "sndfile", - .stream_decode = sndfile_stream_decode, - .scan_file = sndfile_scan_file, - .suffixes = sndfile_suffixes, - .mime_types = sndfile_mime_types, -}; diff --git a/src/decoder/vorbis_comments.c b/src/decoder/vorbis_comments.c deleted file mode 100644 index 6c2d57b72..000000000 --- a/src/decoder/vorbis_comments.c +++ /dev/null @@ -1,156 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "vorbis_comments.h" -#include "tag.h" -#include "tag_table.h" -#include "tag_handler.h" -#include "replay_gain_info.h" - -#include <glib.h> -#include <assert.h> -#include <stddef.h> -#include <string.h> -#include <stdlib.h> - -static const char * -vorbis_comment_value(const char *comment, const char *needle) -{ - size_t len = strlen(needle); - - if (g_ascii_strncasecmp(comment, needle, len) == 0 && - comment[len] == '=') - return comment + len + 1; - - return NULL; -} - -bool -vorbis_comments_to_replay_gain(struct replay_gain_info *rgi, char **comments) -{ - const char *temp; - bool found = false; - - replay_gain_info_init(rgi); - - while (*comments) { - if ((temp = - vorbis_comment_value(*comments, "replaygain_track_gain"))) { - rgi->tuples[REPLAY_GAIN_TRACK].gain = atof(temp); - found = true; - } else if ((temp = vorbis_comment_value(*comments, - "replaygain_album_gain"))) { - rgi->tuples[REPLAY_GAIN_ALBUM].gain = atof(temp); - found = true; - } else if ((temp = vorbis_comment_value(*comments, - "replaygain_track_peak"))) { - rgi->tuples[REPLAY_GAIN_TRACK].peak = atof(temp); - found = true; - } else if ((temp = vorbis_comment_value(*comments, - "replaygain_album_peak"))) { - rgi->tuples[REPLAY_GAIN_ALBUM].peak = atof(temp); - found = true; - } - - comments++; - } - - return found; -} - -/** - * Check if the comment's name equals the passed name, and if so, copy - * the comment value into the tag. - */ -static bool -vorbis_copy_comment(const char *comment, - const char *name, enum tag_type tag_type, - const struct tag_handler *handler, void *handler_ctx) -{ - const char *value; - - value = vorbis_comment_value(comment, name); - if (value != NULL) { - tag_handler_invoke_tag(handler, handler_ctx, tag_type, value); - return true; - } - - return false; -} - -static const struct tag_table vorbis_tags[] = { - { "tracknumber", TAG_TRACK }, - { "discnumber", TAG_DISC }, - { "album artist", TAG_ALBUM_ARTIST }, - { NULL, TAG_NUM_OF_ITEM_TYPES } -}; - -static void -vorbis_scan_comment(const char *comment, - const struct tag_handler *handler, void *handler_ctx) -{ - if (handler->pair != NULL) { - char *name = g_strdup((const char*)comment); - char *value = strchr(name, '='); - - if (value != NULL && value > name) { - *value++ = 0; - tag_handler_invoke_pair(handler, handler_ctx, - name, value); - } - - g_free(name); - } - - for (const struct tag_table *i = vorbis_tags; i->name != NULL; ++i) - if (vorbis_copy_comment(comment, i->name, i->type, - handler, handler_ctx)) - return; - - for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) - if (vorbis_copy_comment(comment, - tag_item_names[i], i, - handler, handler_ctx)) - return; -} - -void -vorbis_comments_scan(char **comments, - const struct tag_handler *handler, void *handler_ctx) -{ - while (*comments) - vorbis_scan_comment(*comments++, - handler, handler_ctx); - -} - -struct tag * -vorbis_comments_to_tag(char **comments) -{ - struct tag *tag = tag_new(); - vorbis_comments_scan(comments, &add_tag_handler, tag); - - if (tag_is_empty(tag)) { - tag_free(tag); - tag = NULL; - } - - return tag; -} diff --git a/src/decoder/vorbis_comments.h b/src/decoder/vorbis_comments.h deleted file mode 100644 index c15096930..000000000 --- a/src/decoder/vorbis_comments.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_VORBIS_COMMENTS_H -#define MPD_VORBIS_COMMENTS_H - -#include "check.h" - -#include <stdbool.h> - -struct replay_gain_info; -struct tag_handler; - -bool -vorbis_comments_to_replay_gain(struct replay_gain_info *rgi, char **comments); - -void -vorbis_comments_scan(char **comments, - const struct tag_handler *handler, void *handler_ctx); - -struct tag * -vorbis_comments_to_tag(char **comments); - -#endif diff --git a/src/decoder/vorbis_decoder_plugin.c b/src/decoder/vorbis_decoder_plugin.c deleted file mode 100644 index 15cdc0ca9..000000000 --- a/src/decoder/vorbis_decoder_plugin.c +++ /dev/null @@ -1,314 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "vorbis_comments.h" -#include "_ogg_common.h" -#include "audio_check.h" -#include "uri.h" -#include "tag_handler.h" - -#ifndef HAVE_TREMOR -#define OV_EXCLUDE_STATIC_CALLBACKS -#include <vorbis/vorbisfile.h> -#else -#include <tremor/ivorbisfile.h> -/* Macros to make Tremor's API look like libogg. Tremor always - returns host-byte-order 16-bit signed data, and uses integer - milliseconds where libogg uses double seconds. -*/ -#define ov_read(VF, BUFFER, LENGTH, BIGENDIANP, WORD, SGNED, BITSTREAM) \ - ov_read(VF, BUFFER, LENGTH, BITSTREAM) -#define ov_time_total(VF, I) ((double)ov_time_total(VF, I)/1000) -#define ov_time_tell(VF) ((double)ov_time_tell(VF)/1000) -#define ov_time_seek_page(VF, S) (ov_time_seek_page(VF, (S)*1000)) -#endif /* HAVE_TREMOR */ - -#include <glib.h> - -#include <assert.h> -#include <errno.h> -#include <unistd.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "vorbis" -#define OGG_CHUNK_SIZE 4096 - -#if G_BYTE_ORDER == G_BIG_ENDIAN -#define OGG_DECODE_USE_BIGENDIAN 1 -#else -#define OGG_DECODE_USE_BIGENDIAN 0 -#endif - -struct vorbis_input_stream { - struct decoder *decoder; - - struct input_stream *input_stream; - bool seekable; -}; - -static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *data) -{ - struct vorbis_input_stream *vis = data; - size_t ret; - - ret = decoder_read(vis->decoder, vis->input_stream, ptr, size * nmemb); - - errno = 0; - - return ret / size; -} - -static int ogg_seek_cb(void *data, ogg_int64_t offset, int whence) -{ - struct vorbis_input_stream *vis = data; - - return vis->seekable && - (!vis->decoder || decoder_get_command(vis->decoder) != DECODE_COMMAND_STOP) && - input_stream_lock_seek(vis->input_stream, offset, whence, NULL) - ? 0 : -1; -} - -/* TODO: check Ogg libraries API and see if we can just not have this func */ -static int ogg_close_cb(G_GNUC_UNUSED void *data) -{ - return 0; -} - -static long ogg_tell_cb(void *data) -{ - const struct vorbis_input_stream *vis = data; - - return (long)vis->input_stream->offset; -} - -static const ov_callbacks vorbis_is_callbacks = { - .read_func = ogg_read_cb, - .seek_func = ogg_seek_cb, - .close_func = ogg_close_cb, - .tell_func = ogg_tell_cb, -}; - -static const char * -vorbis_strerror(int code) -{ - switch (code) { - case OV_EREAD: - return "read error"; - - case OV_ENOTVORBIS: - return "not vorbis stream"; - - case OV_EVERSION: - return "vorbis version mismatch"; - - case OV_EBADHEADER: - return "invalid vorbis header"; - - case OV_EFAULT: - return "internal logic error"; - - default: - return "unknown error"; - } -} - -static bool -vorbis_is_open(struct vorbis_input_stream *vis, OggVorbis_File *vf, - struct decoder *decoder, struct input_stream *input_stream) -{ - vis->decoder = decoder; - vis->input_stream = input_stream; - vis->seekable = input_stream->seekable && - (input_stream->uri == NULL || - !uri_has_scheme(input_stream->uri)); - - int ret = ov_open_callbacks(vis, vf, NULL, 0, vorbis_is_callbacks); - if (ret < 0) { - if (decoder == NULL || - decoder_get_command(decoder) == DECODE_COMMAND_NONE) - g_warning("Failed to open Ogg Vorbis stream: %s", - vorbis_strerror(ret)); - return false; - } - - return true; -} - -static void -vorbis_send_comments(struct decoder *decoder, struct input_stream *is, - char **comments) -{ - struct tag *tag; - - tag = vorbis_comments_to_tag(comments); - if (!tag) - return; - - decoder_tag(decoder, is, tag); - tag_free(tag); -} - -/* public */ -static void -vorbis_stream_decode(struct decoder *decoder, - struct input_stream *input_stream) -{ - GError *error = NULL; - OggVorbis_File vf; - struct vorbis_input_stream vis; - struct audio_format audio_format; - float total_time; - int current_section; - int prev_section = -1; - long ret; - char chunk[OGG_CHUNK_SIZE]; - long bitRate = 0; - long test; - const vorbis_info *vi; - enum decoder_command cmd = DECODE_COMMAND_NONE; - - if (ogg_stream_type_detect(input_stream) != VORBIS) - return; - - /* rewind the stream, because ogg_stream_type_detect() has - moved it */ - input_stream_lock_seek(input_stream, 0, SEEK_SET, NULL); - - if (!vorbis_is_open(&vis, &vf, decoder, input_stream)) - return; - - vi = ov_info(&vf, -1); - if (vi == NULL) { - g_warning("ov_info() has failed"); - return; - } - - if (!audio_format_init_checked(&audio_format, vi->rate, - SAMPLE_FORMAT_S16, - vi->channels, &error)) { - g_warning("%s", error->message); - g_error_free(error); - return; - } - - total_time = ov_time_total(&vf, -1); - if (total_time < 0) - total_time = 0; - - decoder_initialized(decoder, &audio_format, vis.seekable, total_time); - - do { - if (cmd == DECODE_COMMAND_SEEK) { - double seek_where = decoder_seek_where(decoder); - if (0 == ov_time_seek_page(&vf, seek_where)) { - decoder_command_finished(decoder); - } else - decoder_seek_error(decoder); - } - - ret = ov_read(&vf, chunk, sizeof(chunk), - OGG_DECODE_USE_BIGENDIAN, 2, 1, ¤t_section); - if (ret == OV_HOLE) /* bad packet */ - ret = 0; - else if (ret <= 0) - /* break on EOF or other error */ - break; - - if (current_section != prev_section) { - char **comments; - - vi = ov_info(&vf, -1); - if (vi == NULL) { - g_warning("ov_info() has failed"); - break; - } - - if (vi->rate != (long)audio_format.sample_rate || - vi->channels != (int)audio_format.channels) { - /* we don't support audio format - change yet */ - g_warning("audio format change, stopping here"); - break; - } - - comments = ov_comment(&vf, -1)->user_comments; - vorbis_send_comments(decoder, input_stream, comments); - - struct replay_gain_info rgi; - if (vorbis_comments_to_replay_gain(&rgi, comments)) - decoder_replay_gain(decoder, &rgi); - - prev_section = current_section; - } - - if ((test = ov_bitrate_instant(&vf)) > 0) - bitRate = test / 1000; - - cmd = decoder_data(decoder, input_stream, - chunk, ret, - bitRate); - } while (cmd != DECODE_COMMAND_STOP); - - ov_clear(&vf); -} - -static bool -vorbis_scan_stream(struct input_stream *is, - const struct tag_handler *handler, void *handler_ctx) -{ - struct vorbis_input_stream vis; - OggVorbis_File vf; - - if (!vorbis_is_open(&vis, &vf, NULL, is)) - return false; - - tag_handler_invoke_duration(handler, handler_ctx, - (int)(ov_time_total(&vf, -1) + 0.5)); - - vorbis_comments_scan(ov_comment(&vf, -1)->user_comments, - handler, handler_ctx); - - ov_clear(&vf); - return true; -} - -static const char *const vorbis_suffixes[] = { - "ogg", "oga", NULL -}; - -static const char *const vorbis_mime_types[] = { - "application/ogg", - "application/x-ogg", - "audio/ogg", - "audio/vorbis", - "audio/vorbis+ogg", - "audio/x-ogg", - "audio/x-vorbis", - "audio/x-vorbis+ogg", - NULL -}; - -const struct decoder_plugin vorbis_decoder_plugin = { - .name = "vorbis", - .stream_decode = vorbis_stream_decode, - .scan_stream = vorbis_scan_stream, - .suffixes = vorbis_suffixes, - .mime_types = vorbis_mime_types -}; diff --git a/src/decoder/wavpack_decoder_plugin.c b/src/decoder/wavpack_decoder_plugin.c deleted file mode 100644 index 9ebd0fccc..000000000 --- a/src/decoder/wavpack_decoder_plugin.c +++ /dev/null @@ -1,596 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "audio_check.h" -#include "path.h" -#include "utils.h" -#include "tag_table.h" -#include "tag_handler.h" -#include "tag_ape.h" - -#include <wavpack/wavpack.h> -#include <glib.h> - -#include <assert.h> -#include <unistd.h> -#include <stdio.h> -#include <stdlib.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "wavpack" - -#define ERRORLEN 80 - -/** A pointer type for format converter function. */ -typedef void (*format_samples_t)( - int bytes_per_sample, - void *buffer, uint32_t count -); - -/* - * This function has been borrowed from the tiny player found on - * wavpack.com. Modifications were required because mpd only handles - * max 24-bit samples. - */ -static void -format_samples_int(int bytes_per_sample, void *buffer, uint32_t count) -{ - int32_t *src = buffer; - - switch (bytes_per_sample) { - case 1: { - int8_t *dst = buffer; - /* - * The asserts like the following one are because we do the - * formatting of samples within a single buffer. The size - * of the output samples never can be greater than the size - * of the input ones. Otherwise we would have an overflow. - */ - assert_static(sizeof(*dst) <= sizeof(*src)); - - /* pass through and align 8-bit samples */ - while (count--) { - *dst++ = *src++; - } - break; - } - case 2: { - uint16_t *dst = buffer; - assert_static(sizeof(*dst) <= sizeof(*src)); - - /* pass through and align 16-bit samples */ - while (count--) { - *dst++ = *src++; - } - break; - } - - case 3: - case 4: - /* do nothing */ - break; - } -} - -/* - * This function converts floating point sample data to 24-bit integer. - */ -static void -format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer, - uint32_t count) -{ - float *p = buffer; - - while (count--) { - *p /= (1 << 23); - ++p; - } -} - -/** - * Choose a MPD sample format from libwavpacks' number of bits. - */ -static enum sample_format -wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample) -{ - if (is_float) - return SAMPLE_FORMAT_FLOAT; - - switch (bytes_per_sample) { - case 1: - return SAMPLE_FORMAT_S8; - - case 2: - return SAMPLE_FORMAT_S16; - - case 3: - return SAMPLE_FORMAT_S24_P32; - - case 4: - return SAMPLE_FORMAT_S32; - - default: - return SAMPLE_FORMAT_UNDEFINED; - } -} - -/* - * This does the main decoding thing. - * Requires an already opened WavpackContext. - */ -static void -wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) -{ - GError *error = NULL; - bool is_float; - enum sample_format sample_format; - struct audio_format audio_format; - format_samples_t format_samples; - float total_time; - int bytes_per_sample, output_sample_size; - - is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0; - sample_format = - wavpack_bits_to_sample_format(is_float, - WavpackGetBytesPerSample(wpc)); - - if (!audio_format_init_checked(&audio_format, - WavpackGetSampleRate(wpc), - sample_format, - WavpackGetNumChannels(wpc), &error)) { - g_warning("%s", error->message); - g_error_free(error); - return; - } - - if (is_float) { - format_samples = format_samples_float; - } else { - format_samples = format_samples_int; - } - - total_time = WavpackGetNumSamples(wpc); - total_time /= audio_format.sample_rate; - bytes_per_sample = WavpackGetBytesPerSample(wpc); - output_sample_size = audio_format_frame_size(&audio_format); - - /* wavpack gives us all kind of samples in a 32-bit space */ - int32_t chunk[1024]; - const uint32_t samples_requested = G_N_ELEMENTS(chunk) / - audio_format.channels; - - decoder_initialized(decoder, &audio_format, can_seek, total_time); - - enum decoder_command cmd = decoder_get_command(decoder); - while (cmd != DECODE_COMMAND_STOP) { - if (cmd == DECODE_COMMAND_SEEK) { - if (can_seek) { - unsigned where = decoder_seek_where(decoder) * - audio_format.sample_rate; - - if (WavpackSeekSample(wpc, where)) { - decoder_command_finished(decoder); - } else { - decoder_seek_error(decoder); - } - } else { - decoder_seek_error(decoder); - } - } - - uint32_t samples_got = WavpackUnpackSamples(wpc, chunk, - samples_requested); - if (samples_got == 0) - break; - - int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 + - 0.5); - format_samples(bytes_per_sample, chunk, - samples_got * audio_format.channels); - - cmd = decoder_data(decoder, NULL, chunk, - samples_got * output_sample_size, - bitrate); - } -} - -/** - * Locate and parse a floating point tag. Returns true if it was - * found. - */ -static bool -wavpack_tag_float(WavpackContext *wpc, const char *key, float *value_r) -{ - char buffer[64]; - int ret; - - ret = WavpackGetTagItem(wpc, key, buffer, sizeof(buffer)); - if (ret <= 0) - return false; - - *value_r = atof(buffer); - return true; -} - -static bool -wavpack_replaygain(struct replay_gain_info *replay_gain_info, - WavpackContext *wpc) -{ - bool found = false; - - replay_gain_info_init(replay_gain_info); - - found |= wavpack_tag_float( - wpc, "replaygain_track_gain", - &replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain - ); - found |= wavpack_tag_float( - wpc, "replaygain_track_peak", - &replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak - ); - found |= wavpack_tag_float( - wpc, "replaygain_album_gain", - &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain - ); - found |= wavpack_tag_float( - wpc, "replaygain_album_peak", - &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak - ); - - return found; -} - -static void -wavpack_scan_tag_item(WavpackContext *wpc, const char *name, - enum tag_type type, - const struct tag_handler *handler, void *handler_ctx) -{ - char buffer[1024]; - int len = WavpackGetTagItem(wpc, name, buffer, sizeof(buffer)); - if (len <= 0 || (unsigned)len >= sizeof(buffer)) - return; - - tag_handler_invoke_tag(handler, handler_ctx, type, buffer); - -} - -static void -wavpack_scan_pair(WavpackContext *wpc, const char *name, - const struct tag_handler *handler, void *handler_ctx) -{ - char buffer[8192]; - int len = WavpackGetTagItem(wpc, name, buffer, sizeof(buffer)); - if (len <= 0 || (unsigned)len >= sizeof(buffer)) - return; - - tag_handler_invoke_pair(handler, handler_ctx, name, buffer); -} - -/* - * Reads metainfo from the specified file. - */ -static bool -wavpack_scan_file(const char *fname, - const struct tag_handler *handler, void *handler_ctx) -{ - WavpackContext *wpc; - char error[ERRORLEN]; - - wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0); - if (wpc == NULL) { - g_warning( - "failed to open WavPack file \"%s\": %s\n", - fname, error - ); - return false; - } - - tag_handler_invoke_duration(handler, handler_ctx, - WavpackGetNumSamples(wpc) / - WavpackGetSampleRate(wpc)); - - /* the WavPack format implies APEv2 tags, which means we can - reuse the mapping from tag_ape.c */ - - for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) { - const char *name = tag_item_names[i]; - if (name != NULL) - wavpack_scan_tag_item(wpc, name, (enum tag_type)i, - handler, handler_ctx); - } - - for (const struct tag_table *i = ape_tags; i->name != NULL; ++i) - wavpack_scan_tag_item(wpc, i->name, i->type, - handler, handler_ctx); - - if (handler->pair != NULL) { - char name[64]; - - for (int i = 0, n = WavpackGetNumTagItems(wpc); - i < n; ++i) { - int len = WavpackGetTagItemIndexed(wpc, i, name, - sizeof(name)); - if (len <= 0 || (unsigned)len >= sizeof(name)) - continue; - - wavpack_scan_pair(wpc, name, handler, handler_ctx); - } - } - - WavpackCloseFile(wpc); - - return true; -} - -/* - * mpd input_stream <=> WavpackStreamReader wrapper callbacks - */ - -/* This struct is needed for per-stream last_byte storage. */ -struct wavpack_input { - struct decoder *decoder; - struct input_stream *is; - /* Needed for push_back_byte() */ - int last_byte; -}; - -/** - * Little wrapper for struct wavpack_input to cast from void *. - */ -static struct wavpack_input * -wpin(void *id) -{ - assert(id); - return id; -} - -static int32_t -wavpack_input_read_bytes(void *id, void *data, int32_t bcount) -{ - uint8_t *buf = (uint8_t *)data; - int32_t i = 0; - - if (wpin(id)->last_byte != EOF) { - *buf++ = wpin(id)->last_byte; - wpin(id)->last_byte = EOF; - --bcount; - ++i; - } - - /* wavpack fails if we return a partial read, so we just wait - until the buffer is full */ - while (bcount > 0) { - size_t nbytes = decoder_read( - wpin(id)->decoder, wpin(id)->is, buf, bcount - ); - if (nbytes == 0) { - /* EOF, error or a decoder command */ - break; - } - - i += nbytes; - bcount -= nbytes; - buf += nbytes; - } - - return i; -} - -static uint32_t -wavpack_input_get_pos(void *id) -{ - return wpin(id)->is->offset; -} - -static int -wavpack_input_set_pos_abs(void *id, uint32_t pos) -{ - return input_stream_lock_seek(wpin(id)->is, pos, SEEK_SET, NULL) - ? 0 : -1; -} - -static int -wavpack_input_set_pos_rel(void *id, int32_t delta, int mode) -{ - return input_stream_lock_seek(wpin(id)->is, delta, mode, NULL) - ? 0 : -1; -} - -static int -wavpack_input_push_back_byte(void *id, int c) -{ - if (wpin(id)->last_byte == EOF) { - wpin(id)->last_byte = c; - return c; - } else { - return EOF; - } -} - -static uint32_t -wavpack_input_get_length(void *id) -{ - if (wpin(id)->is->size < 0) - return 0; - - return wpin(id)->is->size; -} - -static int -wavpack_input_can_seek(void *id) -{ - return wpin(id)->is->seekable; -} - -static WavpackStreamReader mpd_is_reader = { - .read_bytes = wavpack_input_read_bytes, - .get_pos = wavpack_input_get_pos, - .set_pos_abs = wavpack_input_set_pos_abs, - .set_pos_rel = wavpack_input_set_pos_rel, - .push_back_byte = wavpack_input_push_back_byte, - .get_length = wavpack_input_get_length, - .can_seek = wavpack_input_can_seek, - .write_bytes = NULL /* no need to write edited tags */ -}; - -static void -wavpack_input_init(struct wavpack_input *isp, struct decoder *decoder, - struct input_stream *is) -{ - isp->decoder = decoder; - isp->is = is; - isp->last_byte = EOF; -} - -static struct input_stream * -wavpack_open_wvc(struct decoder *decoder, const char *uri, - GMutex *mutex, GCond *cond, - struct wavpack_input *wpi) -{ - struct input_stream *is_wvc; - char *wvc_url = NULL; - char first_byte; - size_t nbytes; - - /* - * As we use dc->utf8url, this function will be bad for - * single files. utf8url is not absolute file path :/ - */ - if (uri == NULL) - return false; - - wvc_url = g_strconcat(uri, "c", NULL); - is_wvc = input_stream_open(wvc_url, mutex, cond, NULL); - g_free(wvc_url); - - if (is_wvc == NULL) - return NULL; - - /* - * And we try to buffer in order to get know - * about a possible 404 error. - */ - nbytes = decoder_read( - decoder, is_wvc, &first_byte, sizeof(first_byte) - ); - if (nbytes == 0) { - input_stream_close(is_wvc); - return NULL; - } - - /* push it back */ - wavpack_input_init(wpi, decoder, is_wvc); - wpi->last_byte = first_byte; - return is_wvc; -} - -/* - * Decodes a stream. - */ -static void -wavpack_streamdecode(struct decoder * decoder, struct input_stream *is) -{ - char error[ERRORLEN]; - WavpackContext *wpc; - struct input_stream *is_wvc; - int open_flags = OPEN_NORMALIZE; - struct wavpack_input isp, isp_wvc; - bool can_seek = is->seekable; - - is_wvc = wavpack_open_wvc(decoder, is->uri, is->mutex, is->cond, - &isp_wvc); - if (is_wvc != NULL) { - open_flags |= OPEN_WVC; - can_seek &= is_wvc->seekable; - } - - if (!can_seek) { - open_flags |= OPEN_STREAMING; - } - - wavpack_input_init(&isp, decoder, is); - wpc = WavpackOpenFileInputEx( - &mpd_is_reader, &isp, - open_flags & OPEN_WVC ? &isp_wvc : NULL, - error, open_flags, 23 - ); - - if (wpc == NULL) { - g_warning("failed to open WavPack stream: %s\n", error); - return; - } - - wavpack_decode(decoder, wpc, can_seek); - - WavpackCloseFile(wpc); - if (open_flags & OPEN_WVC) { - input_stream_close(is_wvc); - } -} - -/* - * Decodes a file. - */ -static void -wavpack_filedecode(struct decoder *decoder, const char *fname) -{ - char error[ERRORLEN]; - WavpackContext *wpc; - - wpc = WavpackOpenFileInput( - fname, error, - OPEN_TAGS | OPEN_WVC | OPEN_NORMALIZE, 23 - ); - if (wpc == NULL) { - g_warning( - "failed to open WavPack file \"%s\": %s\n", - fname, error - ); - return; - } - - struct replay_gain_info replay_gain_info; - if (wavpack_replaygain(&replay_gain_info, wpc)) - decoder_replay_gain(decoder, &replay_gain_info); - - wavpack_decode(decoder, wpc, true); - - WavpackCloseFile(wpc); -} - -static char const *const wavpack_suffixes[] = { - "wv", - NULL -}; - -static char const *const wavpack_mime_types[] = { - "audio/x-wavpack", - NULL -}; - -const struct decoder_plugin wavpack_decoder_plugin = { - .name = "wavpack", - .stream_decode = wavpack_streamdecode, - .file_decode = wavpack_filedecode, - .scan_file = wavpack_scan_file, - .suffixes = wavpack_suffixes, - .mime_types = wavpack_mime_types -}; diff --git a/src/decoder/wildmidi_decoder_plugin.c b/src/decoder/wildmidi_decoder_plugin.c deleted file mode 100644 index 2cdb30a9c..000000000 --- a/src/decoder/wildmidi_decoder_plugin.c +++ /dev/null @@ -1,150 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "tag_handler.h" -#include "glib_compat.h" - -#include <glib.h> - -#include <wildmidi_lib.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "wildmidi" - -enum { - WILDMIDI_SAMPLE_RATE = 48000, -}; - -static bool -wildmidi_init(const struct config_param *param) -{ - const char *config_file; - int ret; - - config_file = config_get_block_string(param, "config_file", - "/etc/timidity/timidity.cfg"); - if (!g_file_test(config_file, G_FILE_TEST_IS_REGULAR)) { - g_debug("configuration file does not exist: %s", config_file); - return false; - } - - ret = WildMidi_Init(config_file, WILDMIDI_SAMPLE_RATE, 0); - return ret == 0; -} - -static void -wildmidi_finish(void) -{ - WildMidi_Shutdown(); -} - -static void -wildmidi_file_decode(struct decoder *decoder, const char *path_fs) -{ - static const struct audio_format audio_format = { - .sample_rate = WILDMIDI_SAMPLE_RATE, - .format = SAMPLE_FORMAT_S16, - .channels = 2, - }; - midi *wm; - const struct _WM_Info *info; - enum decoder_command cmd; - - wm = WildMidi_Open(path_fs); - if (wm == NULL) - return; - - info = WildMidi_GetInfo(wm); - if (info == NULL) { - WildMidi_Close(wm); - return; - } - - decoder_initialized(decoder, &audio_format, true, - info->approx_total_samples / WILDMIDI_SAMPLE_RATE); - - do { - char buffer[4096]; - int len; - - info = WildMidi_GetInfo(wm); - if (info == NULL) - break; - - len = WildMidi_GetOutput(wm, buffer, sizeof(buffer)); - if (len <= 0) - break; - - cmd = decoder_data(decoder, NULL, buffer, len, 0); - - if (cmd == DECODE_COMMAND_SEEK) { - unsigned long seek_where = WILDMIDI_SAMPLE_RATE * - decoder_seek_where(decoder); - -#ifdef HAVE_WILDMIDI_SAMPLED_SEEK - WildMidi_SampledSeek(wm, &seek_where); -#else - WildMidi_FastSeek(wm, &seek_where); -#endif - decoder_command_finished(decoder); - cmd = DECODE_COMMAND_NONE; - } - - } while (cmd == DECODE_COMMAND_NONE); - - WildMidi_Close(wm); -} - -static bool -wildmidi_scan_file(const char *path_fs, - const struct tag_handler *handler, void *handler_ctx) -{ - midi *wm = WildMidi_Open(path_fs); - if (wm == NULL) - return false; - - const struct _WM_Info *info = WildMidi_GetInfo(wm); - if (info == NULL) { - WildMidi_Close(wm); - return false; - } - - int duration = info->approx_total_samples / WILDMIDI_SAMPLE_RATE; - tag_handler_invoke_duration(handler, handler_ctx, duration); - - WildMidi_Close(wm); - - return true; -} - -static const char *const wildmidi_suffixes[] = { - "mid", - NULL -}; - -const struct decoder_plugin wildmidi_decoder_plugin = { - .name = "wildmidi", - .init = wildmidi_init, - .finish = wildmidi_finish, - .file_decode = wildmidi_file_decode, - .scan_file = wildmidi_scan_file, - .suffixes = wildmidi_suffixes, -}; diff --git a/src/decoder_api.c b/src/decoder_api.c deleted file mode 100644 index a45d0f1e6..000000000 --- a/src/decoder_api.c +++ /dev/null @@ -1,567 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_api.h" -#include "decoder_internal.h" -#include "decoder_control.h" -#include "audio_config.h" -#include "song.h" -#include "buffer.h" -#include "pipe.h" -#include "chunk.h" -#include "replay_gain_config.h" - -#include <glib.h> - -#include <assert.h> -#include <stdlib.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "decoder" - -void -decoder_initialized(struct decoder *decoder, - const struct audio_format *audio_format, - bool seekable, float total_time) -{ - struct decoder_control *dc = decoder->dc; - struct audio_format_string af_string; - - assert(dc->state == DECODE_STATE_START); - assert(dc->pipe != NULL); - assert(decoder != NULL); - assert(decoder->stream_tag == NULL); - assert(decoder->decoder_tag == NULL); - assert(!decoder->seeking); - assert(audio_format != NULL); - assert(audio_format_defined(audio_format)); - assert(audio_format_valid(audio_format)); - - dc->in_audio_format = *audio_format; - getOutputAudioFormat(audio_format, &dc->out_audio_format); - - dc->seekable = seekable; - dc->total_time = total_time; - - decoder_lock(dc); - dc->state = DECODE_STATE_DECODE; - g_cond_signal(dc->client_cond); - decoder_unlock(dc); - - g_debug("audio_format=%s, seekable=%s", - audio_format_to_string(&dc->in_audio_format, &af_string), - seekable ? "true" : "false"); - - if (!audio_format_equals(&dc->in_audio_format, - &dc->out_audio_format)) - g_debug("converting to %s", - audio_format_to_string(&dc->out_audio_format, - &af_string)); -} - -/** - * Checks if we need an "initial seek". If so, then the initial seek - * is prepared, and the function returns true. - */ -G_GNUC_PURE -static bool -decoder_prepare_initial_seek(struct decoder *decoder) -{ - const struct decoder_control *dc = decoder->dc; - assert(dc->pipe != NULL); - - if (dc->state != DECODE_STATE_DECODE) - /* wait until the decoder has finished initialisation - (reading file headers etc.) before emitting the - virtual "SEEK" command */ - return false; - - if (decoder->initial_seek_running) - /* initial seek has already begun - override any other - command */ - return true; - - if (decoder->initial_seek_pending) { - if (!dc->seekable) { - /* seeking is not possible */ - decoder->initial_seek_pending = false; - return false; - } - - if (dc->command == DECODE_COMMAND_NONE) { - /* begin initial seek */ - - decoder->initial_seek_pending = false; - decoder->initial_seek_running = true; - return true; - } - - /* skip initial seek when there's another command - (e.g. STOP) */ - - decoder->initial_seek_pending = false; - } - - return false; -} - -/** - * Returns the current decoder command. May return a "virtual" - * synthesized command, e.g. to seek to the beginning of the CUE - * track. - */ -G_GNUC_PURE -static enum decoder_command -decoder_get_virtual_command(struct decoder *decoder) -{ - const struct decoder_control *dc = decoder->dc; - assert(dc->pipe != NULL); - - if (decoder_prepare_initial_seek(decoder)) - return DECODE_COMMAND_SEEK; - - return dc->command; -} - -enum decoder_command -decoder_get_command(struct decoder *decoder) -{ - return decoder_get_virtual_command(decoder); -} - -void -decoder_command_finished(struct decoder *decoder) -{ - struct decoder_control *dc = decoder->dc; - - decoder_lock(dc); - - assert(dc->command != DECODE_COMMAND_NONE || - decoder->initial_seek_running); - assert(dc->command != DECODE_COMMAND_SEEK || - decoder->initial_seek_running || - dc->seek_error || decoder->seeking); - assert(dc->pipe != NULL); - - if (decoder->initial_seek_running) { - assert(!decoder->seeking); - assert(decoder->chunk == NULL); - assert(music_pipe_empty(dc->pipe)); - - decoder->initial_seek_running = false; - decoder->timestamp = dc->start_ms / 1000.; - decoder_unlock(dc); - return; - } - - if (decoder->seeking) { - decoder->seeking = false; - - /* delete frames from the old song position */ - - if (decoder->chunk != NULL) { - music_buffer_return(dc->buffer, decoder->chunk); - decoder->chunk = NULL; - } - - music_pipe_clear(dc->pipe, dc->buffer); - - decoder->timestamp = dc->seek_where; - } - - dc->command = DECODE_COMMAND_NONE; - g_cond_signal(dc->client_cond); - decoder_unlock(dc); -} - -double decoder_seek_where(G_GNUC_UNUSED struct decoder * decoder) -{ - const struct decoder_control *dc = decoder->dc; - - assert(dc->pipe != NULL); - - if (decoder->initial_seek_running) - return dc->start_ms / 1000.; - - assert(dc->command == DECODE_COMMAND_SEEK); - - decoder->seeking = true; - - return dc->seek_where; -} - -void decoder_seek_error(struct decoder * decoder) -{ - struct decoder_control *dc = decoder->dc; - - assert(dc->pipe != NULL); - - if (decoder->initial_seek_running) { - /* d'oh, we can't seek to the sub-song start position, - what now? - no idea, ignoring the problem for now. */ - decoder->initial_seek_running = false; - return; - } - - assert(dc->command == DECODE_COMMAND_SEEK); - - dc->seek_error = true; - decoder->seeking = false; - - decoder_command_finished(decoder); -} - -/** - * Should be read operation be cancelled? That is the case when the - * player thread has sent a command such as "STOP". - */ -G_GNUC_PURE -static inline bool -decoder_check_cancel_read(const struct decoder *decoder) -{ - if (decoder == NULL) - return false; - - const struct decoder_control *dc = decoder->dc; - if (dc->command == DECODE_COMMAND_NONE) - return false; - - /* ignore the SEEK command during initialization, the plugin - should handle that after it has initialized successfully */ - if (dc->command == DECODE_COMMAND_SEEK && - (dc->state == DECODE_STATE_START || decoder->seeking)) - return false; - - return true; -} - -size_t decoder_read(struct decoder *decoder, - struct input_stream *is, - void *buffer, size_t length) -{ - /* XXX don't allow decoder==NULL */ - GError *error = NULL; - size_t nbytes; - - assert(decoder == NULL || - decoder->dc->state == DECODE_STATE_START || - decoder->dc->state == DECODE_STATE_DECODE); - assert(is != NULL); - assert(buffer != NULL); - - if (length == 0) - return 0; - - input_stream_lock(is); - - while (true) { - if (decoder_check_cancel_read(decoder)) { - input_stream_unlock(is); - return 0; - } - - if (input_stream_available(is)) - break; - - g_cond_wait(is->cond, is->mutex); - } - - nbytes = input_stream_read(is, buffer, length, &error); - assert(nbytes == 0 || error == NULL); - assert(nbytes > 0 || error != NULL || input_stream_eof(is)); - - if (G_UNLIKELY(nbytes == 0 && error != NULL)) { - g_warning("%s", error->message); - g_error_free(error); - } - - input_stream_unlock(is); - - return nbytes; -} - -void -decoder_timestamp(struct decoder *decoder, double t) -{ - assert(decoder != NULL); - assert(t >= 0); - - decoder->timestamp = t; -} - -/** - * Sends a #tag as-is to the music pipe. Flushes the current chunk - * (decoder.chunk) if there is one. - */ -static enum decoder_command -do_send_tag(struct decoder *decoder, const struct tag *tag) -{ - struct music_chunk *chunk; - - if (decoder->chunk != NULL) { - /* there is a partial chunk - flush it, we want the - tag in a new chunk */ - decoder_flush_chunk(decoder); - g_cond_signal(decoder->dc->client_cond); - } - - assert(decoder->chunk == NULL); - - chunk = decoder_get_chunk(decoder); - if (chunk == NULL) { - assert(decoder->dc->command != DECODE_COMMAND_NONE); - return decoder->dc->command; - } - - chunk->tag = tag_dup(tag); - return DECODE_COMMAND_NONE; -} - -static bool -update_stream_tag(struct decoder *decoder, struct input_stream *is) -{ - struct tag *tag; - - tag = is != NULL - ? input_stream_lock_tag(is) - : NULL; - if (tag == NULL) { - tag = decoder->song_tag; - if (tag == NULL) - return false; - - /* no stream tag present - submit the song tag - instead */ - decoder->song_tag = NULL; - } - - if (decoder->stream_tag != NULL) - tag_free(decoder->stream_tag); - - decoder->stream_tag = tag; - return true; -} - -enum decoder_command -decoder_data(struct decoder *decoder, - struct input_stream *is, - const void *_data, size_t length, - uint16_t kbit_rate) -{ - struct decoder_control *dc = decoder->dc; - const char *data = _data; - GError *error = NULL; - enum decoder_command cmd; - - assert(dc->state == DECODE_STATE_DECODE); - assert(dc->pipe != NULL); - assert(length % audio_format_frame_size(&dc->in_audio_format) == 0); - - decoder_lock(dc); - cmd = decoder_get_virtual_command(decoder); - decoder_unlock(dc); - - if (cmd == DECODE_COMMAND_STOP || cmd == DECODE_COMMAND_SEEK || - length == 0) - return cmd; - - /* send stream tags */ - - if (update_stream_tag(decoder, is)) { - if (decoder->decoder_tag != NULL) { - /* merge with tag from decoder plugin */ - struct tag *tag; - - tag = tag_merge(decoder->decoder_tag, - decoder->stream_tag); - cmd = do_send_tag(decoder, tag); - tag_free(tag); - } else - /* send only the stream tag */ - cmd = do_send_tag(decoder, decoder->stream_tag); - - if (cmd != DECODE_COMMAND_NONE) - return cmd; - } - - if (!audio_format_equals(&dc->in_audio_format, &dc->out_audio_format)) { - data = pcm_convert(&decoder->conv_state, - &dc->in_audio_format, data, length, - &dc->out_audio_format, &length, - &error); - if (data == NULL) { - /* the PCM conversion has failed - stop - playback, since we have no better way to - bail out */ - g_warning("%s", error->message); - return DECODE_COMMAND_STOP; - } - } - - while (length > 0) { - struct music_chunk *chunk; - char *dest; - size_t nbytes; - bool full; - - chunk = decoder_get_chunk(decoder); - if (chunk == NULL) { - assert(dc->command != DECODE_COMMAND_NONE); - return dc->command; - } - - dest = music_chunk_write(chunk, &dc->out_audio_format, - decoder->timestamp - - dc->song->start_ms / 1000.0, - kbit_rate, &nbytes); - if (dest == NULL) { - /* the chunk is full, flush it */ - decoder_flush_chunk(decoder); - g_cond_signal(dc->client_cond); - continue; - } - - assert(nbytes > 0); - - if (nbytes > length) - nbytes = length; - - /* copy the buffer */ - - memcpy(dest, data, nbytes); - - /* expand the music pipe chunk */ - - full = music_chunk_expand(chunk, &dc->out_audio_format, nbytes); - if (full) { - /* the chunk is full, flush it */ - decoder_flush_chunk(decoder); - g_cond_signal(dc->client_cond); - } - - data += nbytes; - length -= nbytes; - - decoder->timestamp += (double)nbytes / - audio_format_time_to_size(&dc->out_audio_format); - - if (dc->end_ms > 0 && - decoder->timestamp >= dc->end_ms / 1000.0) - /* the end of this range has been reached: - stop decoding */ - return DECODE_COMMAND_STOP; - } - - return DECODE_COMMAND_NONE; -} - -enum decoder_command -decoder_tag(G_GNUC_UNUSED struct decoder *decoder, struct input_stream *is, - const struct tag *tag) -{ - G_GNUC_UNUSED const struct decoder_control *dc = decoder->dc; - enum decoder_command cmd; - - assert(dc->state == DECODE_STATE_DECODE); - assert(dc->pipe != NULL); - assert(tag != NULL); - - /* save the tag */ - - if (decoder->decoder_tag != NULL) - tag_free(decoder->decoder_tag); - decoder->decoder_tag = tag_dup(tag); - - /* check for a new stream tag */ - - update_stream_tag(decoder, is); - - /* check if we're seeking */ - - if (decoder_prepare_initial_seek(decoder)) - /* during initial seek, no music chunk must be created - until seeking is finished; skip the rest of the - function here */ - return DECODE_COMMAND_SEEK; - - /* send tag to music pipe */ - - if (decoder->stream_tag != NULL) { - /* merge with tag from input stream */ - struct tag *merged; - - merged = tag_merge(decoder->stream_tag, decoder->decoder_tag); - cmd = do_send_tag(decoder, merged); - tag_free(merged); - } else - /* send only the decoder tag */ - cmd = do_send_tag(decoder, tag); - - return cmd; -} - -float -decoder_replay_gain(struct decoder *decoder, - const struct replay_gain_info *replay_gain_info) -{ - float return_db = 0; - assert(decoder != NULL); - - if (replay_gain_info != NULL) { - static unsigned serial; - if (++serial == 0) - serial = 1; - - if (REPLAY_GAIN_OFF != replay_gain_mode) { - return_db = 20.0 * log10f( - replay_gain_tuple_scale( - &replay_gain_info->tuples[replay_gain_get_real_mode()], - replay_gain_preamp, replay_gain_missing_preamp, - replay_gain_limit)); - } - - decoder->replay_gain_info = *replay_gain_info; - decoder->replay_gain_serial = serial; - - if (decoder->chunk != NULL) { - /* flush the current chunk because the new - replay gain values affect the following - samples */ - decoder_flush_chunk(decoder); - g_cond_signal(decoder->dc->client_cond); - } - } else - decoder->replay_gain_serial = 0; - - return return_db; -} - -void -decoder_mixramp(struct decoder *decoder, float replay_gain_db, - char *mixramp_start, char *mixramp_end) -{ - assert(decoder != NULL); - struct decoder_control *dc = decoder->dc; - assert(dc != NULL); - - dc->replay_gain_db = replay_gain_db; - dc_mixramp_start(dc, mixramp_start); - dc_mixramp_end(dc, mixramp_end); -} diff --git a/src/decoder_api.h b/src/decoder_api.h deleted file mode 100644 index 6e011c395..000000000 --- a/src/decoder_api.h +++ /dev/null @@ -1,173 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/*! \file - * \brief The MPD Decoder API - * - * This is the public API which is used by decoder plugins to - * communicate with the mpd core. - */ - -#ifndef MPD_DECODER_API_H -#define MPD_DECODER_API_H - -#include "check.h" -#include "decoder_command.h" -#include "decoder_plugin.h" -#include "input_stream.h" -#include "replay_gain_info.h" -#include "tag.h" -#include "audio_format.h" -#include "conf.h" - -#include <stdbool.h> - -/** - * Notify the player thread that it has finished initialization and - * that it has read the song's meta data. - * - * @param decoder the decoder object - * @param audio_format the audio format which is going to be sent to - * decoder_data() - * @param seekable true if the song is seekable - * @param total_time the total number of seconds in this song; -1 if unknown - */ -void -decoder_initialized(struct decoder *decoder, - const struct audio_format *audio_format, - bool seekable, float total_time); - -/** - * Determines the pending decoder command. - * - * @param decoder the decoder object - * @return the current command, or DECODE_COMMAND_NONE if there is no - * command pending - */ -enum decoder_command -decoder_get_command(struct decoder *decoder); - -/** - * Called by the decoder when it has performed the requested command - * (dc->command). This function resets dc->command and wakes up the - * player thread. - * - * @param decoder the decoder object - */ -void -decoder_command_finished(struct decoder *decoder); - -/** - * Call this when you have received the DECODE_COMMAND_SEEK command. - * - * @param decoder the decoder object - * @return the destination position for the week - */ -double -decoder_seek_where(struct decoder *decoder); - -/** - * Call this instead of decoder_command_finished() when seeking has - * failed. - * - * @param decoder the decoder object - */ -void -decoder_seek_error(struct decoder *decoder); - -/** - * Blocking read from the input stream. - * - * @param decoder the decoder object - * @param is the input stream to read from - * @param buffer the destination buffer - * @param length the maximum number of bytes to read - * @return the number of bytes read, or 0 if one of the following - * occurs: end of file; error; command (like SEEK or STOP). - */ -size_t -decoder_read(struct decoder *decoder, struct input_stream *is, - void *buffer, size_t length); - -/** - * Sets the time stamp for the next data chunk [seconds]. The MPD - * core automatically counts it up, and a decoder plugin only needs to - * use this function if it thinks that adding to the time stamp based - * on the buffer size won't work. - */ -void -decoder_timestamp(struct decoder *decoder, double t); - -/** - * This function is called by the decoder plugin when it has - * successfully decoded block of input data. - * - * @param decoder the decoder object - * @param is an input stream which is buffering while we are waiting - * for the player - * @param data the source buffer - * @param length the number of bytes in the buffer - * @return the current command, or DECODE_COMMAND_NONE if there is no - * command pending - */ -enum decoder_command -decoder_data(struct decoder *decoder, struct input_stream *is, - const void *data, size_t length, - uint16_t kbit_rate); - -/** - * This function is called by the decoder plugin when it has - * successfully decoded a tag. - * - * @param decoder the decoder object - * @param is an input stream which is buffering while we are waiting - * for the player - * @param tag the tag to send - * @return the current command, or DECODE_COMMAND_NONE if there is no - * command pending - */ -enum decoder_command -decoder_tag(struct decoder *decoder, struct input_stream *is, - const struct tag *tag); - -/** - * Set replay gain values for the following chunks. - * - * @param decoder the decoder object - * @param rgi the replay_gain_info object; may be NULL to invalidate - * the previous replay gain values - * @return the replay gain adjustment used - */ -float -decoder_replay_gain(struct decoder *decoder, - const struct replay_gain_info *replay_gain_info); - -/** - * Store MixRamp tags. - * - * @param decoder the decoder object - * @param replay_gain_db the ReplayGain adjustment used for this song - * @param mixramp_start the mixramp_start tag; may be NULL to invalidate - * @param mixramp_end the mixramp_end tag; may be NULL to invalidate - */ -void -decoder_mixramp(struct decoder *decoder, float replay_gain_db, - char *mixramp_start, char *mixramp_end); - -#endif diff --git a/src/decoder_buffer.c b/src/decoder_buffer.c deleted file mode 100644 index fcb135976..000000000 --- a/src/decoder_buffer.c +++ /dev/null @@ -1,167 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_buffer.h" -#include "decoder_api.h" - -#include <glib.h> - -#include <assert.h> - -struct decoder_buffer { - struct decoder *decoder; - struct input_stream *is; - - /** the allocated size of the buffer */ - size_t size; - - /** the current length of the buffer */ - size_t length; - - /** number of bytes already consumed at the beginning of the - buffer */ - size_t consumed; - - /** the actual buffer (dynamic size) */ - unsigned char data[sizeof(size_t)]; -}; - -struct decoder_buffer * -decoder_buffer_new(struct decoder *decoder, struct input_stream *is, - size_t size) -{ - struct decoder_buffer *buffer = - g_malloc(sizeof(*buffer) - sizeof(buffer->data) + size); - - assert(is != NULL); - assert(size > 0); - - buffer->decoder = decoder; - buffer->is = is; - buffer->size = size; - buffer->length = 0; - buffer->consumed = 0; - - return buffer; -} - -void -decoder_buffer_free(struct decoder_buffer *buffer) -{ - assert(buffer != NULL); - - g_free(buffer); -} - -bool -decoder_buffer_is_empty(const struct decoder_buffer *buffer) -{ - return buffer->consumed == buffer->length; -} - -bool -decoder_buffer_is_full(const struct decoder_buffer *buffer) -{ - return buffer->consumed == 0 && buffer->length == buffer->size; -} - -static void -decoder_buffer_shift(struct decoder_buffer *buffer) -{ - assert(buffer->consumed > 0); - - buffer->length -= buffer->consumed; - memmove(buffer->data, buffer->data + buffer->consumed, buffer->length); - buffer->consumed = 0; -} - -bool -decoder_buffer_fill(struct decoder_buffer *buffer) -{ - size_t nbytes; - - if (buffer->consumed > 0) - decoder_buffer_shift(buffer); - - if (buffer->length >= buffer->size) - /* buffer is full */ - return false; - - nbytes = decoder_read(buffer->decoder, buffer->is, - buffer->data + buffer->length, - buffer->size - buffer->length); - if (nbytes == 0) - /* end of file, I/O error or decoder command - received */ - return false; - - buffer->length += nbytes; - assert(buffer->length <= buffer->size); - - return true; -} - -const void * -decoder_buffer_read(const struct decoder_buffer *buffer, size_t *length_r) -{ - if (buffer->consumed >= buffer->length) - /* buffer is empty */ - return NULL; - - *length_r = buffer->length - buffer->consumed; - return buffer->data + buffer->consumed; -} - -void -decoder_buffer_consume(struct decoder_buffer *buffer, size_t nbytes) -{ - /* just move the "consumed" pointer - decoder_buffer_shift() - will do the real work later (called by - decoder_buffer_fill()) */ - buffer->consumed += nbytes; - - assert(buffer->consumed <= buffer->length); -} - -bool -decoder_buffer_skip(struct decoder_buffer *buffer, size_t nbytes) -{ - size_t length; - const void *data; - bool success; - - /* this could probably be optimized by seeking */ - - while (true) { - data = decoder_buffer_read(buffer, &length); - if (data != NULL) { - if (length > nbytes) - length = nbytes; - decoder_buffer_consume(buffer, length); - nbytes -= length; - if (nbytes == 0) - return true; - } - - success = decoder_buffer_fill(buffer); - if (!success) - return false; - } -} diff --git a/src/decoder_buffer.h b/src/decoder_buffer.h deleted file mode 100644 index 77eff5dd1..000000000 --- a/src/decoder_buffer.h +++ /dev/null @@ -1,106 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_BUFFER_H -#define MPD_DECODER_BUFFER_H - -#include <stdbool.h> -#include <stddef.h> - -/** - * This objects handles buffered reads in decoder plugins easily. You - * create a buffer object, and use its high-level methods to fill and - * read it. It will automatically handle shifting the buffer. - */ -struct decoder_buffer; - -struct decoder; -struct input_stream; - -/** - * Creates a new buffer. - * - * @param decoder the decoder object, used for decoder_read(), may be NULL - * @param is the input stream object where we should read from - * @param size the maximum size of the buffer - * @return the new decoder_buffer object - */ -struct decoder_buffer * -decoder_buffer_new(struct decoder *decoder, struct input_stream *is, - size_t size); - -/** - * Frees resources used by the decoder_buffer object. - */ -void -decoder_buffer_free(struct decoder_buffer *buffer); - -bool -decoder_buffer_is_empty(const struct decoder_buffer *buffer); - -bool -decoder_buffer_is_full(const struct decoder_buffer *buffer); - -/** - * Read data from the input_stream and append it to the buffer. - * - * @return true if data was appended; false if there is no data - * available (yet), end of file, I/O error or a decoder command was - * received - */ -bool -decoder_buffer_fill(struct decoder_buffer *buffer); - -/** - * Reads data from the buffer. This data is not yet consumed, you - * have to call decoder_buffer_consume() to do that. The returned - * buffer becomes invalid after a decoder_buffer_fill() or a - * decoder_buffer_consume() call. - * - * @param buffer the decoder_buffer object - * @param length_r pointer to a size_t where you will receive the - * number of bytes available - * @return a pointer to the read buffer, or NULL if there is no data - * available - */ -const void * -decoder_buffer_read(const struct decoder_buffer *buffer, size_t *length_r); - -/** - * Consume (delete, invalidate) a part of the buffer. The "nbytes" - * parameter must not be larger than the length returned by - * decoder_buffer_read(). - * - * @param buffer the decoder_buffer object - * @param nbytes the number of bytes to consume - */ -void -decoder_buffer_consume(struct decoder_buffer *buffer, size_t nbytes); - -/** - * Skips the specified number of bytes, discarding its data. - * - * @param buffer the decoder_buffer object - * @param nbytes the number of bytes to skip - * @return true on success, false on error - */ -bool -decoder_buffer_skip(struct decoder_buffer *buffer, size_t nbytes); - -#endif diff --git a/src/decoder_command.h b/src/decoder_command.h deleted file mode 100644 index 795e13fb2..000000000 --- a/src/decoder_command.h +++ /dev/null @@ -1,30 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_COMMAND_H -#define MPD_DECODER_COMMAND_H - -enum decoder_command { - DECODE_COMMAND_NONE = 0, - DECODE_COMMAND_START, - DECODE_COMMAND_STOP, - DECODE_COMMAND_SEEK -}; - -#endif diff --git a/src/decoder_control.c b/src/decoder_control.c deleted file mode 100644 index 2ce03b666..000000000 --- a/src/decoder_control.c +++ /dev/null @@ -1,190 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_control.h" -#include "pipe.h" - -#include <assert.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "decoder_control" - -struct decoder_control * -dc_new(GCond *client_cond) -{ - struct decoder_control *dc = g_new(struct decoder_control, 1); - - dc->thread = NULL; - - dc->mutex = g_mutex_new(); - dc->cond = g_cond_new(); - dc->client_cond = client_cond; - - dc->state = DECODE_STATE_STOP; - dc->command = DECODE_COMMAND_NONE; - - dc->replay_gain_db = 0; - dc->replay_gain_prev_db = 0; - dc->mixramp_start = NULL; - dc->mixramp_end = NULL; - dc->mixramp_prev_end = NULL; - - return dc; -} - -void -dc_free(struct decoder_control *dc) -{ - g_cond_free(dc->cond); - g_mutex_free(dc->mutex); - g_free(dc->mixramp_start); - g_free(dc->mixramp_end); - g_free(dc->mixramp_prev_end); - g_free(dc); -} - -static void -dc_command_wait_locked(struct decoder_control *dc) -{ - while (dc->command != DECODE_COMMAND_NONE) - g_cond_wait(dc->client_cond, dc->mutex); -} - -static void -dc_command_locked(struct decoder_control *dc, enum decoder_command cmd) -{ - dc->command = cmd; - decoder_signal(dc); - dc_command_wait_locked(dc); -} - -static void -dc_command(struct decoder_control *dc, enum decoder_command cmd) -{ - decoder_lock(dc); - dc_command_locked(dc, cmd); - decoder_unlock(dc); -} - -static void -dc_command_async(struct decoder_control *dc, enum decoder_command cmd) -{ - decoder_lock(dc); - - dc->command = cmd; - decoder_signal(dc); - - decoder_unlock(dc); -} - -void -dc_start(struct decoder_control *dc, struct song *song, - unsigned start_ms, unsigned end_ms, - struct music_buffer *buffer, struct music_pipe *pipe) -{ - assert(song != NULL); - assert(buffer != NULL); - assert(pipe != NULL); - assert(music_pipe_empty(pipe)); - - dc->song = song; - dc->start_ms = start_ms; - dc->end_ms = end_ms; - dc->buffer = buffer; - dc->pipe = pipe; - dc_command(dc, DECODE_COMMAND_START); -} - -void -dc_stop(struct decoder_control *dc) -{ - decoder_lock(dc); - - if (dc->command != DECODE_COMMAND_NONE) - /* Attempt to cancel the current command. If it's too - late and the decoder thread is already executing - the old command, we'll call STOP again in this - function (see below). */ - dc_command_locked(dc, DECODE_COMMAND_STOP); - - if (dc->state != DECODE_STATE_STOP && dc->state != DECODE_STATE_ERROR) - dc_command_locked(dc, DECODE_COMMAND_STOP); - - decoder_unlock(dc); -} - -bool -dc_seek(struct decoder_control *dc, double where) -{ - assert(dc->state != DECODE_STATE_START); - assert(where >= 0.0); - - if (dc->state == DECODE_STATE_STOP || - dc->state == DECODE_STATE_ERROR || !dc->seekable) - return false; - - dc->seek_where = where; - dc->seek_error = false; - dc_command(dc, DECODE_COMMAND_SEEK); - - if (dc->seek_error) - return false; - - return true; -} - -void -dc_quit(struct decoder_control *dc) -{ - assert(dc->thread != NULL); - - dc->quit = true; - dc_command_async(dc, DECODE_COMMAND_STOP); - - g_thread_join(dc->thread); - dc->thread = NULL; -} - -void -dc_mixramp_start(struct decoder_control *dc, char *mixramp_start) -{ - assert(dc != NULL); - - g_free(dc->mixramp_start); - dc->mixramp_start = mixramp_start; -} - -void -dc_mixramp_end(struct decoder_control *dc, char *mixramp_end) -{ - assert(dc != NULL); - - g_free(dc->mixramp_end); - dc->mixramp_end = mixramp_end; -} - -void -dc_mixramp_prev_end(struct decoder_control *dc, char *mixramp_prev_end) -{ - assert(dc != NULL); - - g_free(dc->mixramp_prev_end); - dc->mixramp_prev_end = mixramp_prev_end; -} diff --git a/src/decoder_control.h b/src/decoder_control.h deleted file mode 100644 index 566b153ee..000000000 --- a/src/decoder_control.h +++ /dev/null @@ -1,277 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_CONTROL_H -#define MPD_DECODER_CONTROL_H - -#include "decoder_command.h" -#include "audio_format.h" - -#include <glib.h> - -#include <assert.h> - -enum decoder_state { - DECODE_STATE_STOP = 0, - DECODE_STATE_START, - DECODE_STATE_DECODE, - - /** - * The last "START" command failed, because there was an I/O - * error or because no decoder was able to decode the file. - * This state will only come after START; once the state has - * turned to DECODE, by definition no such error can occur. - */ - DECODE_STATE_ERROR, -}; - -struct decoder_control { - /** the handle of the decoder thread, or NULL if the decoder - thread isn't running */ - GThread *thread; - - /** - * This lock protects #state and #command. - */ - GMutex *mutex; - - /** - * Trigger this object after you have modified #command. This - * is also used by the decoder thread to notify the caller - * when it has finished a command. - */ - GCond *cond; - - /** - * The trigger of this object's client. It is signalled - * whenever an event occurs. - */ - GCond *client_cond; - - enum decoder_state state; - enum decoder_command command; - - bool quit; - bool seek_error; - bool seekable; - double seek_where; - - /** the format of the song file */ - struct audio_format in_audio_format; - - /** the format being sent to the music pipe */ - struct audio_format out_audio_format; - - /** - * The song currently being decoded. This attribute is set by - * the player thread, when it sends the #DECODE_COMMAND_START - * command. - */ - const struct song *song; - - /** - * The initial seek position (in milliseconds), e.g. to the - * start of a sub-track described by a CUE file. - * - * This attribute is set by dc_start(). - */ - unsigned start_ms; - - /** - * The decoder will stop when it reaches this position (in - * milliseconds). 0 means don't stop before the end of the - * file. - * - * This attribute is set by dc_start(). - */ - unsigned end_ms; - - float total_time; - - /** the #music_chunk allocator */ - struct music_buffer *buffer; - - /** - * The destination pipe for decoded chunks. The caller thread - * owns this object, and is responsible for freeing it. - */ - struct music_pipe *pipe; - - float replay_gain_db; - float replay_gain_prev_db; - char *mixramp_start; - char *mixramp_end; - char *mixramp_prev_end; -}; - -G_GNUC_MALLOC -struct decoder_control * -dc_new(GCond *client_cond); - -void -dc_free(struct decoder_control *dc); - -/** - * Locks the #decoder_control object. - */ -static inline void -decoder_lock(struct decoder_control *dc) -{ - g_mutex_lock(dc->mutex); -} - -/** - * Unlocks the #decoder_control object. - */ -static inline void -decoder_unlock(struct decoder_control *dc) -{ - g_mutex_unlock(dc->mutex); -} - -/** - * Waits for a signal on the #decoder_control object. This function - * is only valid in the decoder thread. The object must be locked - * prior to calling this function. - */ -static inline void -decoder_wait(struct decoder_control *dc) -{ - g_cond_wait(dc->cond, dc->mutex); -} - -/** - * Signals the #decoder_control object. This function is only valid - * in the player thread. The object should be locked prior to calling - * this function. - */ -static inline void -decoder_signal(struct decoder_control *dc) -{ - g_cond_signal(dc->cond); -} - -static inline bool -decoder_is_idle(const struct decoder_control *dc) -{ - return dc->state == DECODE_STATE_STOP || - dc->state == DECODE_STATE_ERROR; -} - -static inline bool -decoder_is_starting(const struct decoder_control *dc) -{ - return dc->state == DECODE_STATE_START; -} - -static inline bool -decoder_has_failed(const struct decoder_control *dc) -{ - assert(dc->command == DECODE_COMMAND_NONE); - - return dc->state == DECODE_STATE_ERROR; -} - -static inline bool -decoder_lock_is_idle(struct decoder_control *dc) -{ - bool ret; - - decoder_lock(dc); - ret = decoder_is_idle(dc); - decoder_unlock(dc); - - return ret; -} - -static inline bool -decoder_lock_is_starting(struct decoder_control *dc) -{ - bool ret; - - decoder_lock(dc); - ret = decoder_is_starting(dc); - decoder_unlock(dc); - - return ret; -} - -static inline bool -decoder_lock_has_failed(struct decoder_control *dc) -{ - bool ret; - - decoder_lock(dc); - ret = decoder_has_failed(dc); - decoder_unlock(dc); - - return ret; -} - -static inline const struct song * -decoder_current_song(const struct decoder_control *dc) -{ - switch (dc->state) { - case DECODE_STATE_STOP: - case DECODE_STATE_ERROR: - return NULL; - - case DECODE_STATE_START: - case DECODE_STATE_DECODE: - return dc->song; - } - - assert(false); - return NULL; -} - -/** - * Start the decoder. - * - * @param the decoder - * @param song the song to be decoded - * @param start_ms see #decoder_control - * @param end_ms see #decoder_control - * @param pipe the pipe which receives the decoded chunks (owned by - * the caller) - */ -void -dc_start(struct decoder_control *dc, struct song *song, - unsigned start_ms, unsigned end_ms, - struct music_buffer *buffer, struct music_pipe *pipe); - -void -dc_stop(struct decoder_control *dc); - -bool -dc_seek(struct decoder_control *dc, double where); - -void -dc_quit(struct decoder_control *dc); - -void -dc_mixramp_start(struct decoder_control *dc, char *mixramp_start); - -void -dc_mixramp_end(struct decoder_control *dc, char *mixramp_end); - -void -dc_mixramp_prev_end(struct decoder_control *dc, char *mixramp_prev_end); - -#endif diff --git a/src/decoder_internal.c b/src/decoder_internal.c deleted file mode 100644 index bc349f2ff..000000000 --- a/src/decoder_internal.c +++ /dev/null @@ -1,96 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_internal.h" -#include "decoder_control.h" -#include "pipe.h" -#include "input_stream.h" -#include "buffer.h" -#include "chunk.h" - -#include <assert.h> - -/** - * All chunks are full of decoded data; wait for the player to free - * one. - */ -static enum decoder_command -need_chunks(struct decoder_control *dc, bool do_wait) -{ - if (dc->command == DECODE_COMMAND_STOP || - dc->command == DECODE_COMMAND_SEEK) - return dc->command; - - if (do_wait) { - decoder_wait(dc); - g_cond_signal(dc->client_cond); - - return dc->command; - } - - return DECODE_COMMAND_NONE; -} - -struct music_chunk * -decoder_get_chunk(struct decoder *decoder) -{ - struct decoder_control *dc = decoder->dc; - enum decoder_command cmd; - - assert(decoder != NULL); - - if (decoder->chunk != NULL) - return decoder->chunk; - - do { - decoder->chunk = music_buffer_allocate(dc->buffer); - if (decoder->chunk != NULL) { - decoder->chunk->replay_gain_serial = - decoder->replay_gain_serial; - if (decoder->replay_gain_serial != 0) - decoder->chunk->replay_gain_info = - decoder->replay_gain_info; - - return decoder->chunk; - } - - decoder_lock(dc); - cmd = need_chunks(dc, true); - decoder_unlock(dc); - } while (cmd == DECODE_COMMAND_NONE); - - return NULL; -} - -void -decoder_flush_chunk(struct decoder *decoder) -{ - struct decoder_control *dc = decoder->dc; - - assert(decoder != NULL); - assert(decoder->chunk != NULL); - - if (music_chunk_is_empty(decoder->chunk)) - music_buffer_return(dc->buffer, decoder->chunk); - else - music_pipe_push(dc->pipe, decoder->chunk); - - decoder->chunk = NULL; -} diff --git a/src/decoder_internal.h b/src/decoder_internal.h deleted file mode 100644 index d89e68cfc..000000000 --- a/src/decoder_internal.h +++ /dev/null @@ -1,100 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_INTERNAL_H -#define MPD_DECODER_INTERNAL_H - -#include "decoder_command.h" -#include "pcm_convert.h" -#include "replay_gain_info.h" - -struct input_stream; - -struct decoder { - struct decoder_control *dc; - - struct pcm_convert_state conv_state; - - /** - * The time stamp of the next data chunk, in seconds. - */ - double timestamp; - - /** - * Is the initial seek (to the start position of the sub-song) - * pending, or has it been performed already? - */ - bool initial_seek_pending; - - /** - * Is the initial seek currently running? During this time, - * the decoder command is SEEK. This flag is set by - * decoder_get_virtual_command(), when the virtual SEEK - * command is generated for the first time. - */ - bool initial_seek_running; - - /** - * This flag is set by decoder_seek_where(), and checked by - * decoder_command_finished(). It is used to clean up after - * seeking. - */ - bool seeking; - - /** - * The tag from the song object. This is only used for local - * files, because we expect the stream server to send us a new - * tag each time we play it. - */ - struct tag *song_tag; - - /** the last tag received from the stream */ - struct tag *stream_tag; - - /** the last tag received from the decoder plugin */ - struct tag *decoder_tag; - - /** the chunk currently being written to */ - struct music_chunk *chunk; - - struct replay_gain_info replay_gain_info; - - /** - * A positive serial number for checking if replay gain info - * has changed since the last check. - */ - unsigned replay_gain_serial; -}; - -/** - * Returns the current chunk the decoder writes to, or allocates a new - * chunk if there is none. - * - * @return the chunk, or NULL if we have received a decoder command - */ -struct music_chunk * -decoder_get_chunk(struct decoder *decoder); - -/** - * Flushes the current chunk. - */ -void -decoder_flush_chunk(struct decoder *decoder); - -#endif diff --git a/src/decoder_list.c b/src/decoder_list.c deleted file mode 100644 index 177b632ad..000000000 --- a/src/decoder_list.c +++ /dev/null @@ -1,235 +0,0 @@ -/* - * Copyright (C) 2003-2012 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_list.h" -#include "decoder_plugin.h" -#include "utils.h" -#include "conf.h" -#include "mpd_error.h" -#include "decoder/pcm_decoder_plugin.h" -#include "decoder/dsdiff_decoder_plugin.h" -#include "decoder/dsf_decoder_plugin.h" - -#include <glib.h> - -#include <string.h> - -extern const struct decoder_plugin mad_decoder_plugin; -extern const struct decoder_plugin mpg123_decoder_plugin; -extern const struct decoder_plugin vorbis_decoder_plugin; -extern const struct decoder_plugin flac_decoder_plugin; -extern const struct decoder_plugin oggflac_decoder_plugin; -extern const struct decoder_plugin sndfile_decoder_plugin; -extern const struct decoder_plugin audiofile_decoder_plugin; -extern const struct decoder_plugin mp4ff_decoder_plugin; -extern const struct decoder_plugin faad_decoder_plugin; -extern const struct decoder_plugin mpcdec_decoder_plugin; -extern const struct decoder_plugin wavpack_decoder_plugin; -extern const struct decoder_plugin modplug_decoder_plugin; -extern const struct decoder_plugin mikmod_decoder_plugin; -extern const struct decoder_plugin sidplay_decoder_plugin; -extern const struct decoder_plugin wildmidi_decoder_plugin; -extern const struct decoder_plugin fluidsynth_decoder_plugin; -extern const struct decoder_plugin ffmpeg_decoder_plugin; -extern const struct decoder_plugin gme_decoder_plugin; - -const struct decoder_plugin *const decoder_plugins[] = { -#ifdef HAVE_MAD - &mad_decoder_plugin, -#endif -#ifdef HAVE_MPG123 - &mpg123_decoder_plugin, -#endif -#ifdef ENABLE_VORBIS_DECODER - &vorbis_decoder_plugin, -#endif -#if defined(HAVE_FLAC) - &oggflac_decoder_plugin, -#endif -#ifdef HAVE_FLAC - &flac_decoder_plugin, -#endif -#ifdef ENABLE_SNDFILE - &sndfile_decoder_plugin, -#endif -#ifdef HAVE_AUDIOFILE - &audiofile_decoder_plugin, -#endif - &dsdiff_decoder_plugin, - &dsf_decoder_plugin, -#ifdef HAVE_FAAD - &faad_decoder_plugin, -#endif -#ifdef HAVE_MP4 - &mp4ff_decoder_plugin, -#endif -#ifdef HAVE_MPCDEC - &mpcdec_decoder_plugin, -#endif -#ifdef HAVE_WAVPACK - &wavpack_decoder_plugin, -#endif -#ifdef HAVE_MODPLUG - &modplug_decoder_plugin, -#endif -#ifdef ENABLE_MIKMOD_DECODER - &mikmod_decoder_plugin, -#endif -#ifdef ENABLE_SIDPLAY - &sidplay_decoder_plugin, -#endif -#ifdef ENABLE_WILDMIDI - &wildmidi_decoder_plugin, -#endif -#ifdef ENABLE_FLUIDSYNTH - &fluidsynth_decoder_plugin, -#endif -#ifdef HAVE_FFMPEG - &ffmpeg_decoder_plugin, -#endif -#ifdef HAVE_GME - &gme_decoder_plugin, -#endif - &pcm_decoder_plugin, - NULL -}; - -enum { - num_decoder_plugins = G_N_ELEMENTS(decoder_plugins) - 1, -}; - -/** which plugins have been initialized successfully? */ -bool decoder_plugins_enabled[num_decoder_plugins]; - -static unsigned -decoder_plugin_index(const struct decoder_plugin *plugin) -{ - unsigned i = 0; - - while (decoder_plugins[i] != plugin) - ++i; - - return i; -} - -static unsigned -decoder_plugin_next_index(const struct decoder_plugin *plugin) -{ - return plugin == 0 - ? 0 /* start with first plugin */ - : decoder_plugin_index(plugin) + 1; -} - -const struct decoder_plugin * -decoder_plugin_from_suffix(const char *suffix, - const struct decoder_plugin *plugin) -{ - if (suffix == NULL) - return NULL; - - for (unsigned i = decoder_plugin_next_index(plugin); - decoder_plugins[i] != NULL; ++i) { - plugin = decoder_plugins[i]; - if (decoder_plugins_enabled[i] && - decoder_plugin_supports_suffix(plugin, suffix)) - return plugin; - } - - return NULL; -} - -const struct decoder_plugin * -decoder_plugin_from_mime_type(const char *mimeType, unsigned int next) -{ - static unsigned i = num_decoder_plugins; - - if (mimeType == NULL) - return NULL; - - if (!next) - i = 0; - for (; decoder_plugins[i] != NULL; ++i) { - const struct decoder_plugin *plugin = decoder_plugins[i]; - if (decoder_plugins_enabled[i] && - decoder_plugin_supports_mime_type(plugin, mimeType)) { - ++i; - return plugin; - } - } - - return NULL; -} - -const struct decoder_plugin * -decoder_plugin_from_name(const char *name) -{ - decoder_plugins_for_each_enabled(plugin) - if (strcmp(plugin->name, name) == 0) - return plugin; - - return NULL; -} - -/** - * Find the "decoder" configuration block for the specified plugin. - * - * @param plugin_name the name of the decoder plugin - * @return the configuration block, or NULL if none was configured - */ -static const struct config_param * -decoder_plugin_config(const char *plugin_name) -{ - const struct config_param *param = NULL; - - while ((param = config_get_next_param(CONF_DECODER, param)) != NULL) { - const char *name = - config_get_block_string(param, "plugin", NULL); - if (name == NULL) - MPD_ERROR("decoder configuration without 'plugin' name in line %d", - param->line); - - if (strcmp(name, plugin_name) == 0) - return param; - } - - return NULL; -} - -void decoder_plugin_init_all(void) -{ - for (unsigned i = 0; decoder_plugins[i] != NULL; ++i) { - const struct decoder_plugin *plugin = decoder_plugins[i]; - const struct config_param *param = - decoder_plugin_config(plugin->name); - - if (!config_get_block_bool(param, "enabled", true)) - /* the plugin is disabled in mpd.conf */ - continue; - - if (decoder_plugin_init(plugin, param)) - decoder_plugins_enabled[i] = true; - } -} - -void decoder_plugin_deinit_all(void) -{ - decoder_plugins_for_each_enabled(plugin) - decoder_plugin_finish(plugin); -} diff --git a/src/decoder_list.h b/src/decoder_list.h deleted file mode 100644 index d0a6ade7e..000000000 --- a/src/decoder_list.h +++ /dev/null @@ -1,65 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_LIST_H -#define MPD_DECODER_LIST_H - -#include <stdbool.h> - -struct decoder_plugin; - -extern const struct decoder_plugin *const decoder_plugins[]; -extern bool decoder_plugins_enabled[]; - -#define decoder_plugins_for_each(plugin) \ - for (const struct decoder_plugin *plugin, \ - *const*decoder_plugin_iterator = &decoder_plugins[0]; \ - (plugin = *decoder_plugin_iterator) != NULL; \ - ++decoder_plugin_iterator) - -#define decoder_plugins_for_each_enabled(plugin) \ - decoder_plugins_for_each(plugin) \ - if (decoder_plugins_enabled[decoder_plugin_iterator - decoder_plugins]) - -/* interface for using plugins */ - -/** - * Find the next enabled decoder plugin which supports the specified suffix. - * - * @param suffix the file name suffix - * @param plugin the previous plugin, or NULL to find the first plugin - * @return a plugin, or NULL if none matches - */ -const struct decoder_plugin * -decoder_plugin_from_suffix(const char *suffix, - const struct decoder_plugin *plugin); - -const struct decoder_plugin * -decoder_plugin_from_mime_type(const char *mimeType, unsigned int next); - -const struct decoder_plugin * -decoder_plugin_from_name(const char *name); - -/* this is where we "load" all the "plugins" ;-) */ -void decoder_plugin_init_all(void); - -/* this is where we "unload" all the "plugins" */ -void decoder_plugin_deinit_all(void); - -#endif diff --git a/src/decoder_plugin.c b/src/decoder_plugin.c deleted file mode 100644 index d32043f0e..000000000 --- a/src/decoder_plugin.c +++ /dev/null @@ -1,47 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_plugin.h" -#include "string_util.h" - -#include <assert.h> - -bool -decoder_plugin_supports_suffix(const struct decoder_plugin *plugin, - const char *suffix) -{ - assert(plugin != NULL); - assert(suffix != NULL); - - return plugin->suffixes != NULL && - string_array_contains(plugin->suffixes, suffix); - -} - -bool -decoder_plugin_supports_mime_type(const struct decoder_plugin *plugin, - const char *mime_type) -{ - assert(plugin != NULL); - assert(mime_type != NULL); - - return plugin->mime_types != NULL && - string_array_contains(plugin->mime_types, mime_type); -} diff --git a/src/decoder_plugin.h b/src/decoder_plugin.h deleted file mode 100644 index 933ba6751..000000000 --- a/src/decoder_plugin.h +++ /dev/null @@ -1,207 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_PLUGIN_H -#define MPD_DECODER_PLUGIN_H - -#include <stdbool.h> -#include <stddef.h> - -struct config_param; -struct input_stream; -struct tag; -struct tag_handler; - -/** - * Opaque handle which the decoder plugin passes to the functions in - * this header. - */ -struct decoder; - -struct decoder_plugin { - const char *name; - - /** - * Initialize the decoder plugin. Optional method. - * - * @param param a configuration block for this plugin, or NULL - * if none is configured - * @return true if the plugin was initialized successfully, - * false if the plugin is not available - */ - bool (*init)(const struct config_param *param); - - /** - * Deinitialize a decoder plugin which was initialized - * successfully. Optional method. - */ - void (*finish)(void); - - /** - * Decode a stream (data read from an #input_stream object). - * - * Either implement this method or file_decode(). If - * possible, it is recommended to implement this method, - * because it is more versatile. - */ - void (*stream_decode)(struct decoder *decoder, - struct input_stream *is); - - /** - * Decode a local file. - * - * Either implement this method or stream_decode(). - */ - void (*file_decode)(struct decoder *decoder, const char *path_fs); - - /** - * Scan metadata of a file. - * - * @return false if the operation has failed - */ - bool (*scan_file)(const char *path_fs, - const struct tag_handler *handler, - void *handler_ctx); - - /** - * Scan metadata of a file. - * - * @return false if the operation has failed - */ - bool (*scan_stream)(struct input_stream *is, - const struct tag_handler *handler, - void *handler_ctx); - - /** - * @brief Return a "virtual" filename for subtracks in - * container formats like flac - * @param const char* pathname full pathname for the file on fs - * @param const unsigned int tnum track number - * - * @return NULL if there are no multiple files - * a filename for every single track according to tnum (param 2) - * do not include full pathname here, just the "virtual" file - */ - char* (*container_scan)(const char *path_fs, const unsigned int tnum); - - /* last element in these arrays must always be a NULL: */ - const char *const*suffixes; - const char *const*mime_types; -}; - -/** - * Initialize a decoder plugin. - * - * @param param a configuration block for this plugin, or NULL if none - * is configured - * @return true if the plugin was initialized successfully, false if - * the plugin is not available - */ -static inline bool -decoder_plugin_init(const struct decoder_plugin *plugin, - const struct config_param *param) -{ - return plugin->init != NULL - ? plugin->init(param) - : true; -} - -/** - * Deinitialize a decoder plugin which was initialized successfully. - */ -static inline void -decoder_plugin_finish(const struct decoder_plugin *plugin) -{ - if (plugin->finish != NULL) - plugin->finish(); -} - -/** - * Decode a stream. - */ -static inline void -decoder_plugin_stream_decode(const struct decoder_plugin *plugin, - struct decoder *decoder, struct input_stream *is) -{ - plugin->stream_decode(decoder, is); -} - -/** - * Decode a file. - */ -static inline void -decoder_plugin_file_decode(const struct decoder_plugin *plugin, - struct decoder *decoder, const char *path_fs) -{ - plugin->file_decode(decoder, path_fs); -} - -/** - * Read the tag of a file. - */ -static inline bool -decoder_plugin_scan_file(const struct decoder_plugin *plugin, - const char *path_fs, - const struct tag_handler *handler, void *handler_ctx) -{ - return plugin->scan_file != NULL - ? plugin->scan_file(path_fs, handler, handler_ctx) - : false; -} - -/** - * Read the tag of a stream. - */ -static inline bool -decoder_plugin_scan_stream(const struct decoder_plugin *plugin, - struct input_stream *is, - const struct tag_handler *handler, - void *handler_ctx) -{ - return plugin->scan_stream != NULL - ? plugin->scan_stream(is, handler, handler_ctx) - : false; -} - -/** - * return "virtual" tracks in a container - */ -static inline char * -decoder_plugin_container_scan( const struct decoder_plugin *plugin, - const char* pathname, - const unsigned int tnum) -{ - return plugin->container_scan(pathname, tnum); -} - -/** - * Does the plugin announce the specified file name suffix? - */ -bool -decoder_plugin_supports_suffix(const struct decoder_plugin *plugin, - const char *suffix); - -/** - * Does the plugin announce the specified MIME type? - */ -bool -decoder_plugin_supports_mime_type(const struct decoder_plugin *plugin, - const char *mime_type); - -#endif diff --git a/src/decoder_print.c b/src/decoder_print.c deleted file mode 100644 index e14477ed8..000000000 --- a/src/decoder_print.c +++ /dev/null @@ -1,53 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_print.h" -#include "decoder_list.h" -#include "decoder_plugin.h" -#include "client.h" - -#include <assert.h> - -static void -decoder_plugin_print(struct client *client, - const struct decoder_plugin *plugin) -{ - const char *const*p; - - assert(plugin != NULL); - assert(plugin->name != NULL); - - client_printf(client, "plugin: %s\n", plugin->name); - - if (plugin->suffixes != NULL) - for (p = plugin->suffixes; *p != NULL; ++p) - client_printf(client, "suffix: %s\n", *p); - - if (plugin->mime_types != NULL) - for (p = plugin->mime_types; *p != NULL; ++p) - client_printf(client, "mime_type: %s\n", *p); -} - -void -decoder_list_print(struct client *client) -{ - decoder_plugins_for_each_enabled(plugin) - decoder_plugin_print(client, plugin); -} diff --git a/src/decoder_print.h b/src/decoder_print.h deleted file mode 100644 index 31713d5d8..000000000 --- a/src/decoder_print.h +++ /dev/null @@ -1,28 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_PRINT_H -#define MPD_DECODER_PRINT_H - -struct client; - -void -decoder_list_print(struct client *client); - -#endif diff --git a/src/decoder_thread.c b/src/decoder_thread.c deleted file mode 100644 index af80ed45b..000000000 --- a/src/decoder_thread.c +++ /dev/null @@ -1,510 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "config.h" -#include "decoder_thread.h" -#include "decoder_control.h" -#include "decoder_internal.h" -#include "decoder_list.h" -#include "decoder_plugin.h" -#include "decoder_api.h" -#include "replay_gain_ape.h" -#include "input_stream.h" -#include "pipe.h" -#include "song.h" -#include "tag.h" -#include "mapper.h" -#include "path.h" -#include "uri.h" -#include "mpd_error.h" - -#include <glib.h> - -#include <unistd.h> -#include <stdio.h> /* for SEEK_SET */ - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "decoder_thread" - -/** - * Marks the current decoder command as "finished" and notifies the - * player thread. - * - * @param dc the #decoder_control object; must be locked - */ -static void -decoder_command_finished_locked(struct decoder_control *dc) -{ - assert(dc->command != DECODE_COMMAND_NONE); - - dc->command = DECODE_COMMAND_NONE; - - g_cond_signal(dc->client_cond); -} - -/** - * Opens the input stream with input_stream_open(), and waits until - * the stream gets ready. If a decoder STOP command is received - * during that, it cancels the operation (but does not close the - * stream). - * - * Unlock the decoder before calling this function. - * - * @return an input_stream on success or if #DECODE_COMMAND_STOP is - * received, NULL on error - */ -static struct input_stream * -decoder_input_stream_open(struct decoder_control *dc, const char *uri) -{ - GError *error = NULL; - struct input_stream *is; - - is = input_stream_open(uri, dc->mutex, dc->cond, &error); - if (is == NULL) { - if (error != NULL) { - g_warning("%s", error->message); - g_error_free(error); - } - - return NULL; - } - - /* wait for the input stream to become ready; its metadata - will be available then */ - - decoder_lock(dc); - - input_stream_update(is); - while (!is->ready && - dc->command != DECODE_COMMAND_STOP) { - decoder_wait(dc); - - input_stream_update(is); - } - - if (!input_stream_check(is, &error)) { - decoder_unlock(dc); - - g_warning("%s", error->message); - g_error_free(error); - - return NULL; - } - - decoder_unlock(dc); - - return is; -} - -static bool -decoder_stream_decode(const struct decoder_plugin *plugin, - struct decoder *decoder, - struct input_stream *input_stream) -{ - assert(plugin != NULL); - assert(plugin->stream_decode != NULL); - assert(decoder != NULL); - assert(decoder->stream_tag == NULL); - assert(decoder->decoder_tag == NULL); - assert(input_stream != NULL); - assert(input_stream->ready); - assert(decoder->dc->state == DECODE_STATE_START); - - g_debug("probing plugin %s", plugin->name); - - if (decoder->dc->command == DECODE_COMMAND_STOP) - return true; - - /* rewind the stream, so each plugin gets a fresh start */ - input_stream_seek(input_stream, 0, SEEK_SET, NULL); - - decoder_unlock(decoder->dc); - - decoder_plugin_stream_decode(plugin, decoder, input_stream); - - decoder_lock(decoder->dc); - - assert(decoder->dc->state == DECODE_STATE_START || - decoder->dc->state == DECODE_STATE_DECODE); - - return decoder->dc->state != DECODE_STATE_START; -} - -static bool -decoder_file_decode(const struct decoder_plugin *plugin, - struct decoder *decoder, const char *path) -{ - assert(plugin != NULL); - assert(plugin->file_decode != NULL); - assert(decoder != NULL); - assert(decoder->stream_tag == NULL); - assert(decoder->decoder_tag == NULL); - assert(path != NULL); - assert(g_path_is_absolute(path)); - assert(decoder->dc->state == DECODE_STATE_START); - - g_debug("probing plugin %s", plugin->name); - - if (decoder->dc->command == DECODE_COMMAND_STOP) - return true; - - decoder_unlock(decoder->dc); - - decoder_plugin_file_decode(plugin, decoder, path); - - decoder_lock(decoder->dc); - - assert(decoder->dc->state == DECODE_STATE_START || - decoder->dc->state == DECODE_STATE_DECODE); - - return decoder->dc->state != DECODE_STATE_START; -} - -/** - * Hack to allow tracking const decoder plugins in a GSList. - */ -static inline gpointer -deconst_plugin(const struct decoder_plugin *plugin) -{ - union { - const struct decoder_plugin *in; - gpointer out; - } u = { .in = plugin }; - - return u.out; -} - -/** - * Try decoding a stream, using plugins matching the stream's MIME type. - * - * @param tried_r a list of plugins which were tried - */ -static bool -decoder_run_stream_mime_type(struct decoder *decoder, struct input_stream *is, - GSList **tried_r) -{ - assert(tried_r != NULL); - - const struct decoder_plugin *plugin; - unsigned int next = 0; - - if (is->mime == NULL) - return false; - - while ((plugin = decoder_plugin_from_mime_type(is->mime, next++))) { - if (plugin->stream_decode == NULL) - continue; - - if (g_slist_find(*tried_r, plugin) != NULL) - /* don't try a plugin twice */ - continue; - - if (decoder_stream_decode(plugin, decoder, is)) - return true; - - *tried_r = g_slist_prepend(*tried_r, deconst_plugin(plugin)); - } - - return false; -} - -/** - * Try decoding a stream, using plugins matching the stream's URI - * suffix. - * - * @param tried_r a list of plugins which were tried - */ -static bool -decoder_run_stream_suffix(struct decoder *decoder, struct input_stream *is, - const char *uri, GSList **tried_r) -{ - assert(tried_r != NULL); - - const char *suffix = uri_get_suffix(uri); - const struct decoder_plugin *plugin = NULL; - - if (suffix == NULL) - return false; - - while ((plugin = decoder_plugin_from_suffix(suffix, plugin)) != NULL) { - if (plugin->stream_decode == NULL) - continue; - - if (g_slist_find(*tried_r, plugin) != NULL) - /* don't try a plugin twice */ - continue; - - if (decoder_stream_decode(plugin, decoder, is)) - return true; - - *tried_r = g_slist_prepend(*tried_r, deconst_plugin(plugin)); - } - - return false; -} - -/** - * Try decoding a stream, using the fallback plugin. - */ -static bool -decoder_run_stream_fallback(struct decoder *decoder, struct input_stream *is) -{ - const struct decoder_plugin *plugin; - - plugin = decoder_plugin_from_name("mad"); - return plugin != NULL && plugin->stream_decode != NULL && - decoder_stream_decode(plugin, decoder, is); -} - -/** - * Try decoding a stream. - */ -static bool -decoder_run_stream(struct decoder *decoder, const char *uri) -{ - struct decoder_control *dc = decoder->dc; - struct input_stream *input_stream; - bool success; - - decoder_unlock(dc); - - input_stream = decoder_input_stream_open(dc, uri); - if (input_stream == NULL) { - decoder_lock(dc); - return false; - } - - decoder_lock(dc); - - GSList *tried = NULL; - - success = dc->command == DECODE_COMMAND_STOP || - /* first we try mime types: */ - decoder_run_stream_mime_type(decoder, input_stream, &tried) || - /* if that fails, try suffix matching the URL: */ - decoder_run_stream_suffix(decoder, input_stream, uri, - &tried) || - /* fallback to mp3: this is needed for bastard streams - that don't have a suffix or set the mimeType */ - (tried == NULL && - decoder_run_stream_fallback(decoder, input_stream)); - - g_slist_free(tried); - - decoder_unlock(dc); - input_stream_close(input_stream); - decoder_lock(dc); - - return success; -} - -/** - * Attempt to load replay gain data, and pass it to - * decoder_replay_gain(). - */ -static void -decoder_load_replay_gain(struct decoder *decoder, const char *path_fs) -{ - struct replay_gain_info info; - if (replay_gain_ape_read(path_fs, &info)) - decoder_replay_gain(decoder, &info); -} - -/** - * Try decoding a file. - */ -static bool -decoder_run_file(struct decoder *decoder, const char *path_fs) -{ - struct decoder_control *dc = decoder->dc; - const char *suffix = uri_get_suffix(path_fs); - const struct decoder_plugin *plugin = NULL; - - if (suffix == NULL) - return false; - - decoder_unlock(dc); - - decoder_load_replay_gain(decoder, path_fs); - - while ((plugin = decoder_plugin_from_suffix(suffix, plugin)) != NULL) { - if (plugin->file_decode != NULL) { - decoder_lock(dc); - - if (decoder_file_decode(plugin, decoder, path_fs)) - return true; - - decoder_unlock(dc); - } else if (plugin->stream_decode != NULL) { - struct input_stream *input_stream; - bool success; - - input_stream = decoder_input_stream_open(dc, path_fs); - if (input_stream == NULL) - continue; - - decoder_lock(dc); - - success = decoder_stream_decode(plugin, decoder, - input_stream); - - decoder_unlock(dc); - - input_stream_close(input_stream); - - if (success) { - decoder_lock(dc); - return true; - } - } - } - - decoder_lock(dc); - return false; -} - -static void -decoder_run_song(struct decoder_control *dc, - const struct song *song, const char *uri) -{ - struct decoder decoder = { - .dc = dc, - .initial_seek_pending = dc->start_ms > 0, - .initial_seek_running = false, - }; - int ret; - - decoder.timestamp = 0.0; - decoder.seeking = false; - decoder.song_tag = song->tag != NULL && song_is_file(song) - ? tag_dup(song->tag) : NULL; - decoder.stream_tag = NULL; - decoder.decoder_tag = NULL; - decoder.chunk = NULL; - - dc->state = DECODE_STATE_START; - - decoder_command_finished_locked(dc); - - pcm_convert_init(&decoder.conv_state); - - ret = song_is_file(song) - ? decoder_run_file(&decoder, uri) - : decoder_run_stream(&decoder, uri); - - decoder_unlock(dc); - - pcm_convert_deinit(&decoder.conv_state); - - /* flush the last chunk */ - - if (decoder.chunk != NULL) - decoder_flush_chunk(&decoder); - - if (decoder.song_tag != NULL) - tag_free(decoder.song_tag); - - if (decoder.stream_tag != NULL) - tag_free(decoder.stream_tag); - - if (decoder.decoder_tag != NULL) - tag_free(decoder.decoder_tag); - - decoder_lock(dc); - - dc->state = ret ? DECODE_STATE_STOP : DECODE_STATE_ERROR; -} - -static void -decoder_run(struct decoder_control *dc) -{ - const struct song *song = dc->song; - char *uri; - - assert(song != NULL); - - if (song_is_file(song)) - uri = map_song_fs(song); - else - uri = song_get_uri(song); - - if (uri == NULL) { - dc->state = DECODE_STATE_ERROR; - decoder_command_finished_locked(dc); - return; - } - - decoder_run_song(dc, song, uri); - g_free(uri); - -} - -static gpointer -decoder_task(gpointer arg) -{ - struct decoder_control *dc = arg; - - decoder_lock(dc); - - do { - assert(dc->state == DECODE_STATE_STOP || - dc->state == DECODE_STATE_ERROR); - - switch (dc->command) { - case DECODE_COMMAND_START: - dc_mixramp_start(dc, NULL); - dc_mixramp_prev_end(dc, dc->mixramp_end); - dc->mixramp_end = NULL; /* Don't free, it's copied above. */ - dc->replay_gain_prev_db = dc->replay_gain_db; - dc->replay_gain_db = 0; - - /* fall through */ - - case DECODE_COMMAND_SEEK: - decoder_run(dc); - break; - - case DECODE_COMMAND_STOP: - decoder_command_finished_locked(dc); - break; - - case DECODE_COMMAND_NONE: - decoder_wait(dc); - break; - } - } while (dc->command != DECODE_COMMAND_NONE || !dc->quit); - - decoder_unlock(dc); - - return NULL; -} - -void -decoder_thread_start(struct decoder_control *dc) -{ - GError *e = NULL; - - assert(dc->thread == NULL); - - dc->quit = false; - - dc->thread = g_thread_create(decoder_task, dc, true, &e); - if (dc->thread == NULL) - MPD_ERROR("Failed to spawn decoder task: %s", e->message); -} diff --git a/src/decoder_thread.h b/src/decoder_thread.h deleted file mode 100644 index 78f12a54a..000000000 --- a/src/decoder_thread.h +++ /dev/null @@ -1,28 +0,0 @@ -/* - * Copyright (C) 2003-2011 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#ifndef MPD_DECODER_THREAD_H -#define MPD_DECODER_THREAD_H - -struct decoder_control; - -void -decoder_thread_start(struct decoder_control *dc); - -#endif |