From c412d6251e9cd3abe735b7622af4003502e54f72 Mon Sep 17 00:00:00 2001
From: Max Kellermann <max@duempel.org>
Date: Tue, 10 Nov 2009 17:11:34 +0100
Subject: audio_format: changed "bits" to "enum sample_format"
This patch prepares support for floating point samples (and probably
other formats). It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.
---
src/audio_check.c | 4 +-
src/audio_check.h | 6 +--
src/audio_format.c | 29 +++++++++++-
src/audio_format.h | 86 ++++++++++++++++++++++++++++--------
src/audio_parser.c | 42 +++++++++++++++---
src/decoder/_flac_common.c | 27 ++++++++++-
src/decoder/_flac_common.h | 2 +
src/decoder/audiofile_plugin.c | 26 +++++++++--
src/decoder/faad_plugin.c | 2 +-
src/decoder/ffmpeg_plugin.c | 30 +++++++++----
src/decoder/flac_pcm.c | 16 ++++---
src/decoder/flac_pcm.h | 4 +-
src/decoder/fluidsynth_plugin.c | 2 +-
src/decoder/mad_plugin.c | 3 +-
src/decoder/mikmod_plugin.c | 2 +-
src/decoder/modplug_plugin.c | 2 +-
src/decoder/mp4ff_plugin.c | 3 +-
src/decoder/mpcdec_plugin.c | 3 +-
src/decoder/mpg123_decoder_plugin.c | 2 +-
src/decoder/sidplay_plugin.cxx | 2 +-
src/decoder/sndfile_decoder_plugin.c | 3 +-
src/decoder/vorbis_plugin.c | 3 +-
src/decoder/wavpack_plugin.c | 47 +++++++++++++++-----
src/decoder/wildmidi_plugin.c | 2 +-
src/encoder/flac_encoder.c | 44 +++++++++++++-----
src/encoder/lame_encoder.c | 2 +-
src/encoder/twolame_encoder.c | 2 +-
src/encoder/vorbis_encoder.c | 2 +-
src/encoder/wave_encoder.c | 29 ++++++++++--
src/filter/volume_filter_plugin.c | 5 ++-
src/normalize.c | 3 +-
src/output/alsa_plugin.c | 73 ++++++++++++++++++------------
src/output/ao_plugin.c | 23 +++++++---
src/output/jack_output_plugin.c | 11 ++---
src/output/mvp_plugin.c | 12 ++---
src/output/openal_plugin.c | 20 +++++----
src/output/oss_plugin.c | 9 ++--
src/output/osx_plugin.c | 19 ++++++--
src/output/pulse_output_plugin.c | 2 +-
src/output/solaris_output_plugin.c | 4 +-
src/pcm_convert.c | 36 +++++++--------
src/pcm_format.c | 45 +++++++++++--------
src/pcm_format.h | 8 ++--
src/pcm_mix.c | 11 ++---
src/pcm_volume.c | 8 ++--
45 files changed, 506 insertions(+), 210 deletions(-)
(limited to 'src')
diff --git a/src/audio_check.c b/src/audio_check.c
index 4046e8862..a843975fa 100644
--- a/src/audio_check.c
+++ b/src/audio_check.c
@@ -35,7 +35,7 @@ audio_check_sample_rate(unsigned long sample_rate, GError **error_r)
}
bool
-audio_check_sample_format(unsigned sample_format, GError **error_r)
+audio_check_sample_format(enum sample_format sample_format, GError **error_r)
{
if (!audio_valid_sample_format(sample_format)) {
g_set_error(error_r, audio_format_quark(), 0,
@@ -60,7 +60,7 @@ audio_check_channel_count(unsigned channels, GError **error_r)
bool
audio_format_init_checked(struct audio_format *af, unsigned long sample_rate,
- unsigned sample_format, unsigned channels,
+ enum sample_format sample_format, unsigned channels,
GError **error_r)
{
if (audio_check_sample_rate(sample_rate, error_r) &&
diff --git a/src/audio_check.h b/src/audio_check.h
index a11fbe3d3..197dedd48 100644
--- a/src/audio_check.h
+++ b/src/audio_check.h
@@ -20,11 +20,11 @@
#ifndef MPD_AUDIO_CHECK_H
#define MPD_AUDIO_CHECK_H
+#include "audio_format.h"
+
#include <glib.h>
#include <stdbool.h>
-struct audio_format;
-
/**
* The GLib quark used for errors reported by this library.
*/
@@ -48,7 +48,7 @@ audio_check_channel_count(unsigned sample_format, GError **error_r);
*/
bool
audio_format_init_checked(struct audio_format *af, unsigned long sample_rate,
- unsigned sample_format, unsigned channels,
+ enum sample_format sample_format, unsigned channels,
GError **error_r);
#endif
diff --git a/src/audio_format.c b/src/audio_format.c
index f88735c73..33cd90f58 100644
--- a/src/audio_format.c
+++ b/src/audio_format.c
@@ -28,6 +28,31 @@
#define REVERSE_ENDIAN_SUFFIX "_be"
#endif
+const char *
+sample_format_to_string(enum sample_format format)
+{
+ switch (format) {
+ case SAMPLE_FORMAT_UNDEFINED:
+ return "?";
+
+ case SAMPLE_FORMAT_S8:
+ return "8";
+
+ case SAMPLE_FORMAT_S16:
+ return "16";
+
+ case SAMPLE_FORMAT_S24_P32:
+ return "24";
+
+ case SAMPLE_FORMAT_S32:
+ return "32";
+ }
+
+ /* unreachable */
+ assert(false);
+ return "?";
+}
+
const char *
audio_format_to_string(const struct audio_format *af,
struct audio_format_string *s)
@@ -35,8 +60,8 @@ audio_format_to_string(const struct audio_format *af,
assert(af != NULL);
assert(s != NULL);
- snprintf(s->buffer, sizeof(s->buffer), "%u:%u%s:%u",
- af->sample_rate, af->bits,
+ snprintf(s->buffer, sizeof(s->buffer), "%u:%s%s:%u",
+ af->sample_rate, sample_format_to_string(af->format),
af->reverse_endian ? REVERSE_ENDIAN_SUFFIX : "",
af->channels);
diff --git a/src/audio_format.h b/src/audio_format.h
index 0c1e425a9..6e7d50c46 100644
--- a/src/audio_format.h
+++ b/src/audio_format.h
@@ -23,6 +23,21 @@
#include <stdint.h>
#include <stdbool.h>
+enum sample_format {
+ SAMPLE_FORMAT_UNDEFINED = 0,
+
+ SAMPLE_FORMAT_S8,
+ SAMPLE_FORMAT_S16,
+
+ /**
+ * Signed 24 bit integer samples, packed in 32 bit integers
+ * (the most significant byte is filled with the sign bit).
+ */
+ SAMPLE_FORMAT_S24_P32,
+
+ SAMPLE_FORMAT_S32,
+};
+
/**
* This structure describes the format of a raw PCM stream.
*/
@@ -35,11 +50,10 @@ struct audio_format {
uint32_t sample_rate;
/**
- * The number of significant bits per sample. Samples are
- * currently always signed. Supported values are 8, 16, 24,
- * 32. 24 bit samples are packed in 32 bit integers.
+ * The format samples are stored in. See the #sample_format
+ * enum for valid values.
*/
- uint8_t bits;
+ uint8_t format;
/**
* The number of channels. Only mono (1) and stereo (2) are
@@ -69,7 +83,7 @@ struct audio_format_string {
static inline void audio_format_clear(struct audio_format *af)
{
af->sample_rate = 0;
- af->bits = 0;
+ af->format = SAMPLE_FORMAT_UNDEFINED;
af->channels = 0;
af->reverse_endian = 0;
}
@@ -80,10 +94,10 @@ static inline void audio_format_clear(struct audio_format *af)
*/
static inline void audio_format_init(struct audio_format *af,
uint32_t sample_rate,
- uint8_t bits, uint8_t channels)
+ enum sample_format format, uint8_t channels)
{
af->sample_rate = sample_rate;
- af->bits = bits;
+ af->format = (uint8_t)format;
af->channels = channels;
af->reverse_endian = 0;
}
@@ -105,7 +119,8 @@ static inline bool audio_format_defined(const struct audio_format *af)
static inline bool
audio_format_fully_defined(const struct audio_format *af)
{
- return af->sample_rate != 0 && af->bits != 0 && af->channels != 0;
+ return af->sample_rate != 0 && af->format != SAMPLE_FORMAT_UNDEFINED &&
+ af->channels != 0;
}
/**
@@ -115,7 +130,8 @@ audio_format_fully_defined(const struct audio_format *af)
static inline bool
audio_format_mask_defined(const struct audio_format *af)
{
- return af->sample_rate != 0 || af->bits != 0 || af->channels != 0;
+ return af->sample_rate != 0 || af->format != SAMPLE_FORMAT_UNDEFINED ||
+ af->channels != 0;
}
/**
@@ -135,9 +151,20 @@ audio_valid_sample_rate(unsigned sample_rate)
* @param bits the number of significant bits per sample
*/
static inline bool
-audio_valid_sample_format(unsigned bits)
+audio_valid_sample_format(enum sample_format format)
{
- return bits == 16 || bits == 24 || bits == 32 || bits == 8;
+ switch (format) {
+ case SAMPLE_FORMAT_S8:
+ case SAMPLE_FORMAT_S16:
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
+ return true;
+
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+ }
+
+ return false;
}
/**
@@ -156,7 +183,7 @@ audio_valid_channel_count(unsigned channels)
static inline bool audio_format_valid(const struct audio_format *af)
{
return audio_valid_sample_rate(af->sample_rate) &&
- audio_valid_sample_format(af->bits) &&
+ audio_valid_sample_format((enum sample_format)af->format) &&
audio_valid_channel_count(af->channels);
}
@@ -168,7 +195,8 @@ static inline bool audio_format_mask_valid(const struct audio_format *af)
{
return (af->sample_rate == 0 ||
audio_valid_sample_rate(af->sample_rate)) &&
- (af->bits == 0 || audio_valid_sample_format(af->bits)) &&
+ (af->format == SAMPLE_FORMAT_UNDEFINED ||
+ audio_valid_sample_format((enum sample_format)af->format)) &&
(af->channels == 0 || audio_valid_channel_count(af->channels));
}
@@ -176,7 +204,7 @@ static inline bool audio_format_equals(const struct audio_format *a,
const struct audio_format *b)
{
return a->sample_rate == b->sample_rate &&
- a->bits == b->bits &&
+ a->format == b->format &&
a->channels == b->channels &&
a->reverse_endian == b->reverse_endian;
}
@@ -188,8 +216,8 @@ audio_format_mask_apply(struct audio_format *af,
if (mask->sample_rate != 0)
af->sample_rate = mask->sample_rate;
- if (mask->bits != 0)
- af->bits = mask->bits;
+ if (mask->format != SAMPLE_FORMAT_UNDEFINED)
+ af->format = mask->format;
if (mask->channels != 0)
af->channels = mask->channels;
@@ -200,12 +228,22 @@ audio_format_mask_apply(struct audio_format *af,
*/
static inline unsigned audio_format_sample_size(const struct audio_format *af)
{
- if (af->bits <= 8)
+ switch (af->format) {
+ case SAMPLE_FORMAT_S8:
return 1;
- else if (af->bits <= 16)
+
+ case SAMPLE_FORMAT_S16:
return 2;
- else
+
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
return 4;
+
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+ }
+
+ return 0;
}
/**
@@ -226,6 +264,16 @@ static inline double audio_format_time_to_size(const struct audio_format *af)
return af->sample_rate * audio_format_frame_size(af);
}
+/**
+ * Renders a #sample_format enum into a string, e.g. for printing it
+ * in a log file.
+ *
+ * @param format a #sample_format enum value
+ * @return the string
+ */
+const char *
+sample_format_to_string(enum sample_format format);
+
/**
* Renders the #audio_format object into a string, e.g. for printing
* it in a log file.
diff --git a/src/audio_parser.c b/src/audio_parser.c
index df87be325..210ea7a62 100644
--- a/src/audio_parser.c
+++ b/src/audio_parser.c
@@ -27,6 +27,7 @@
#include "audio_format.h"
#include "audio_check.h"
+#include <assert.h>
#include <stdlib.h>
/**
@@ -65,14 +66,16 @@ parse_sample_rate(const char *src, bool mask, uint32_t *sample_rate_r,
}
static bool
-parse_sample_format(const char *src, bool mask, uint8_t *bits_r,
+parse_sample_format(const char *src, bool mask,
+ enum sample_format *sample_format_r,
const char **endptr_r, GError **error_r)
{
unsigned long value;
char *endptr;
+ enum sample_format sample_format;
if (mask && *src == '*') {
- *bits_r = 0;
+ *sample_format_r = SAMPLE_FORMAT_UNDEFINED;
*endptr_r = src + 1;
return true;
}
@@ -82,10 +85,34 @@ parse_sample_format(const char *src, bool mask, uint8_t *bits_r,
g_set_error(error_r, audio_parser_quark(), 0,
"Failed to parse the sample format");
return false;
- } else if (!audio_check_sample_format(value, error_r))
+ }
+
+ switch (value) {
+ case 8:
+ sample_format = SAMPLE_FORMAT_S8;
+ break;
+
+ case 16:
+ sample_format = SAMPLE_FORMAT_S16;
+ break;
+
+ case 24:
+ sample_format = SAMPLE_FORMAT_S24_P32;
+ break;
+
+ case 32:
+ sample_format = SAMPLE_FORMAT_S32;
+ break;
+
+ default:
+ g_set_error(error_r, audio_parser_quark(), 0,
+ "Invalid sample format: %lu", value);
return false;
+ }
+
+ assert(audio_valid_sample_format(sample_format));
- *bits_r = value;
+ *sample_format_r = sample_format;
*endptr_r = endptr;
return true;
}
@@ -121,7 +148,8 @@ audio_format_parse(struct audio_format *dest, const char *src,
bool mask, GError **error_r)
{
uint32_t rate;
- uint8_t bits, channels;
+ enum sample_format sample_format;
+ uint8_t channels;
audio_format_clear(dest);
@@ -138,7 +166,7 @@ audio_format_parse(struct audio_format *dest, const char *src,
/* parse sample format */
- if (!parse_sample_format(src, mask, &bits, &src, error_r))
+ if (!parse_sample_format(src, mask, &sample_format, &src, error_r))
return false;
if (*src++ != ':') {
@@ -158,7 +186,7 @@ audio_format_parse(struct audio_format *dest, const char *src,
return false;
}
- audio_format_init(dest, rate, bits, channels);
+ audio_format_init(dest, rate, sample_format, channels);
return true;
}
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index f12b8bff0..70b2c0202 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -60,6 +60,27 @@ flac_data_deinit(struct flac_data *data)
tag_free(data->tag);
}
+static enum sample_format
+flac_sample_format(const FLAC__StreamMetadata_StreamInfo *si)
+{
+ switch (si->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;
+ }
+}
+
bool
flac_data_get_audio_format(struct flac_data *data,
struct audio_format *audio_format)
@@ -71,9 +92,11 @@ flac_data_get_audio_format(struct flac_data *data,
return false;
}
+ data->sample_format = flac_sample_format(&data->stream_info);
+
if (!audio_format_init_checked(audio_format,
data->stream_info.sample_rate,
- data->stream_info.bits_per_sample,
+ data->sample_format,
data->stream_info.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
@@ -144,7 +167,7 @@ flac_common_write(struct flac_data *data, const FLAC__Frame * frame,
buffer = pcm_buffer_get(&data->buffer, buffer_size);
flac_convert(buffer, frame->header.channels,
- frame->header.bits_per_sample, buf,
+ data->sample_format, buf,
0, frame->header.blocksize);
if (data->next_frame >= data->first_frame)
diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h
index 1d211fcfb..2f328afa6 100644
--- a/src/decoder/_flac_common.h
+++ b/src/decoder/_flac_common.h
@@ -38,6 +38,8 @@
struct flac_data {
struct pcm_buffer buffer;
+ enum sample_format sample_format;
+
/**
* The size of one frame in the output buffer.
*/
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index 5a2096d00..fcb431db7 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -101,13 +101,33 @@ setup_virtual_fops(struct input_stream *stream)
return vf;
}
-static uint8_t
+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(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;
@@ -117,7 +137,7 @@ audiofile_setup_sample_format(AFfilehandle af_fp)
AF_SAMPFMT_TWOSCOMP, bits);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- return bits;
+ return audiofile_bits_to_sample_format(bits);
}
static void
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index 55df15555..2a05e33e8 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -283,7 +283,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
decoder_buffer_consume(buffer, nbytes);
return audio_format_init_checked(audio_format, sample_rate,
- 16, channels, error_r);
+ SAMPLE_FORMAT_S16, channels, error_r);
}
/**
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index 9b025153b..af9320322 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -277,6 +277,26 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
return cmd;
}
+static enum sample_format
+ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
+{
+#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
+ int bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+
+ /* XXX implement & test other sample formats */
+
+ switch (bits) {
+ case 16:
+ return SAMPLE_FORMAT_S16;
+ }
+
+ return SAMPLE_FORMAT_UNDEFINED;
+#else
+ /* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
+ return SAMPLE_FORMAT_S16;
+#endif
+}
+
static bool
ffmpeg_decode_internal(struct ffmpeg_context *ctx)
{
@@ -288,7 +308,6 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
struct audio_format audio_format;
enum decoder_command cmd;
int total_time;
- uint8_t bits;
total_time = 0;
@@ -296,14 +315,9 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
codec_context->channels = 2;
}
-#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
- bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
-#else
- /* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
- bits = (uint8_t) 16;
-#endif
if (!audio_format_init_checked(&audio_format,
- codec_context->sample_rate, bits,
+ codec_context->sample_rate,
+ ffmpeg_sample_format(codec_context),
codec_context->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c
index 737d5b043..a8bf6f293 100644
--- a/src/decoder/flac_pcm.c
+++ b/src/decoder/flac_pcm.c
@@ -20,6 +20,8 @@
#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)
@@ -74,12 +76,12 @@ flac_convert_8(int8_t *dest,
void
flac_convert(void *dest,
- unsigned int num_channels, unsigned sample_format,
+ unsigned int num_channels, enum sample_format sample_format,
const FLAC__int32 *const buf[],
unsigned int position, unsigned int end)
{
switch (sample_format) {
- case 16:
+ case SAMPLE_FORMAT_S16:
if (num_channels == 2)
flac_convert_stereo16((int16_t*)dest, buf,
position, end);
@@ -88,15 +90,19 @@ flac_convert(void *dest,
position, end);
break;
- case 24:
- case 32:
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
flac_convert_32((int32_t*)dest, num_channels, buf,
position, end);
break;
- case 8:
+ case SAMPLE_FORMAT_S8:
flac_convert_8((int8_t*)dest, num_channels, buf,
position, end);
break;
+
+ case SAMPLE_FORMAT_UNDEFINED:
+ /* unreachable */
+ assert(false);
}
}
diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h
index dca9d6824..4d7a51c4d 100644
--- a/src/decoder/flac_pcm.h
+++ b/src/decoder/flac_pcm.h
@@ -20,11 +20,13 @@
#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, unsigned sample_format,
+ unsigned int num_channels, enum sample_format sample_format,
const FLAC__int32 *const buf[],
unsigned int position, unsigned int end);
diff --git a/src/decoder/fluidsynth_plugin.c b/src/decoder/fluidsynth_plugin.c
index 3e8a4edc4..1b1e6a531 100644
--- a/src/decoder/fluidsynth_plugin.c
+++ b/src/decoder/fluidsynth_plugin.c
@@ -88,7 +88,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
{
static const struct audio_format audio_format = {
.sample_rate = 48000,
- .bits = 16,
+ .format = SAMPLE_FORMAT_S16,
.channels = 2,
};
char setting_sample_rate[] = "synth.sample-rate";
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index da93fe45b..cba40aea0 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -1188,7 +1188,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
}
if (!audio_format_init_checked(&audio_format,
- data.frame.header.samplerate, 24,
+ data.frame.header.samplerate,
+ SAMPLE_FORMAT_S24_P32,
MAD_NCHANNELS(&data.frame.header),
&error)) {
g_warning("%s", error->message);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 204dd5ce0..1e64aeffd 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -163,7 +163,7 @@ mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs)
/* Prevent module from looping forever */
handle->loop = 0;
- audio_format_init(&audio_format, mikmod_sample_rate, 16, 2);
+ 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);
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index 6c08c2199..02292992d 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -122,7 +122,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
return;
}
- audio_format_init(&audio_format, 44100, 16, 2);
+ audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
assert(audio_format_valid(&audio_format));
decoder_initialized(decoder, &audio_format,
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index 70ca4bdc3..8d3a4b9e9 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -132,7 +132,8 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
return NULL;
}
- if (!audio_format_init_checked(audio_format, sample_rate, 16, channels,
+ if (!audio_format_init_checked(audio_format, sample_rate,
+ SAMPLE_FORMAT_S16, channels,
&error)) {
g_warning("%s", error->message);
g_error_free(error);
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index d985f8459..2f1936e55 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -196,7 +196,8 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
mpc_demux_get_info(demux, &info);
#endif
- if (!audio_format_init_checked(&audio_format, info.sample_freq, 24,
+ 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);
diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c
index 922e56484..62e6b00b0 100644
--- a/src/decoder/mpg123_decoder_plugin.c
+++ b/src/decoder/mpg123_decoder_plugin.c
@@ -87,7 +87,7 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
return false;
}
- if (!audio_format_init_checked(audio_format, rate, 16,
+ if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16,
channels, &gerror)) {
g_warning("%s", gerror->message);
g_error_free(gerror);
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index b6e557e08..63e46a285 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -277,7 +277,7 @@ 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, 16, 2);
+ audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, 2);
assert(audio_format_valid(&audio_format));
decoder_initialized(decoder, &audio_format, true, (float)song_len);
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
index 84835c449..fee0f8292 100644
--- a/src/decoder/sndfile_decoder_plugin.c
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -130,7 +130,8 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
/* 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, 32,
+ if (!audio_format_init_checked(&audio_format, info.samplerate,
+ SAMPLE_FORMAT_S32,
info.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index 3a41869a0..cb61e5999 100755
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -311,7 +311,8 @@ vorbis_stream_decode(struct decoder *decoder,
return;
}
- if (!audio_format_init_checked(&audio_format, vi->rate, 16,
+ if (!audio_format_init_checked(&audio_format, vi->rate,
+ SAMPLE_FORMAT_S16,
vi->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 9b32a79f2..dffa078f9 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -123,6 +123,33 @@ format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer,
}
}
+/**
+ * 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_S24_P32;
+
+ 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.
@@ -132,7 +159,8 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
struct replay_gain_info *replay_gain_info)
{
GError *error = NULL;
- unsigned bits;
+ bool is_float;
+ enum sample_format sample_format;
struct audio_format audio_format;
format_samples_t format_samples;
char chunk[CHUNK_SIZE];
@@ -141,19 +169,14 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
int bytes_per_sample, output_sample_size;
int position;
- bits = WavpackGetBitsPerSample(wpc);
-
- /* round bitwidth to 8-bit units */
- bits = (bits + 7) & (~7);
- /* MPD handles max 32-bit samples */
- if (bits > 32)
- bits = 32;
-
- if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
- bits = 24;
+ 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), bits,
+ WavpackGetSampleRate(wpc),
+ sample_format,
WavpackGetNumChannels(wpc), &error)) {
g_warning("%s", error->message);
g_error_free(error);
diff --git a/src/decoder/wildmidi_plugin.c b/src/decoder/wildmidi_plugin.c
index 718f24c2e..70b4d5ef9 100644
--- a/src/decoder/wildmidi_plugin.c
+++ b/src/decoder/wildmidi_plugin.c
@@ -59,7 +59,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
{
static const struct audio_format audio_format = {
.sample_rate = WILDMIDI_SAMPLE_RATE,
- .bits = 16,
+ .format = SAMPLE_FORMAT_S16,
.channels = 2,
};
midi *wm;
diff --git a/src/encoder/flac_encoder.c b/src/encoder/flac_encoder.c
index ab7dc0c39..4f80fe568 100644
--- a/src/encoder/flac_encoder.c
+++ b/src/encoder/flac_encoder.c
@@ -89,7 +89,8 @@ flac_encoder_finish(struct encoder *_encoder)
}
static bool
-flac_encoder_setup(struct flac_encoder *encoder, GError **error)
+flac_encoder_setup(struct flac_encoder *encoder, unsigned bits_per_sample,
+ GError **error)
{
if ( !FLAC__stream_encoder_set_compression_level(encoder->fse,
encoder->compression)) {
@@ -106,10 +107,10 @@ flac_encoder_setup(struct flac_encoder *encoder, GError **error)
return false;
}
if ( !FLAC__stream_encoder_set_bits_per_sample(encoder->fse,
- encoder->audio_format.bits)) {
+ bits_per_sample)) {
g_set_error(error, flac_encoder_quark(), 0,
"error setting flac bit format to %d",
- encoder->audio_format.bits);
+ bits_per_sample);
return false;
}
if ( !FLAC__stream_encoder_set_sample_rate(encoder->fse,
@@ -143,13 +144,29 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
GError **error)
{
struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+ unsigned bits_per_sample;
FLAC__StreamEncoderInitStatus init_status;
encoder->audio_format = *audio_format;
/* FIXME: flac should support 32bit as well */
- if (audio_format->bits > 24)
- audio_format->bits = 24;
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S8:
+ bits_per_sample = 8;
+ break;
+
+ case SAMPLE_FORMAT_S16:
+ bits_per_sample = 16;
+ break;
+
+ case SAMPLE_FORMAT_S24_P32:
+ bits_per_sample = 24;
+ break;
+
+ default:
+ bits_per_sample = 24;
+ audio_format->format = SAMPLE_FORMAT_S24_P32;
+ }
/* allocate the encoder */
encoder->fse = FLAC__stream_encoder_new();
@@ -159,7 +176,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
return false;
}
- if (!flac_encoder_setup(encoder, error)) {
+ if (!flac_encoder_setup(encoder, bits_per_sample, error)) {
FLAC__stream_encoder_delete(encoder->fse);
return false;
}
@@ -237,20 +254,23 @@ flac_encoder_write(struct encoder *_encoder,
num_frames = length / audio_format_frame_size(&encoder->audio_format);
num_samples = num_frames * encoder->audio_format.channels;
- switch (encoder->audio_format.bits) {
- case 8:
+ switch (encoder->audio_format.format) {
+ case SAMPLE_FORMAT_S8:
exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*4);
pcm8_to_flac(exbuffer, data, num_samples);
buffer = exbuffer;
break;
- case 16:
+
+ case SAMPLE_FORMAT_S16:
exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*2);
pcm16_to_flac(exbuffer, data, num_samples);
buffer = exbuffer;
break;
- case 24:
- case 32: /* nothing need to be done
- * format is the same for both mpd and libFLAC */
+
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
+ /* nothing need to be done; format is the same for
+ both mpd and libFLAC */
buffer = data;
break;
}
diff --git a/src/encoder/lame_encoder.c b/src/encoder/lame_encoder.c
index 812ff39c5..97431a817 100644
--- a/src/encoder/lame_encoder.c
+++ b/src/encoder/lame_encoder.c
@@ -185,7 +185,7 @@ lame_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
{
struct lame_encoder *encoder = (struct lame_encoder *)_encoder;
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
audio_format->channels = 2;
encoder->audio_format = *audio_format;
diff --git a/src/encoder/twolame_encoder.c b/src/encoder/twolame_encoder.c
index cddf5773e..e7af89bf6 100644
--- a/src/encoder/twolame_encoder.c
+++ b/src/encoder/twolame_encoder.c
@@ -192,7 +192,7 @@ twolame_encoder_open(struct encoder *_encoder, struct audio_format *audio_format
{
struct twolame_encoder *encoder = (struct twolame_encoder *)_encoder;
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
audio_format->channels = 2;
encoder->audio_format = *audio_format;
diff --git a/src/encoder/vorbis_encoder.c b/src/encoder/vorbis_encoder.c
index 2fa0fd950..d072bcd3f 100644
--- a/src/encoder/vorbis_encoder.c
+++ b/src/encoder/vorbis_encoder.c
@@ -212,7 +212,7 @@ vorbis_encoder_open(struct encoder *_encoder,
struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
bool ret;
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
encoder->audio_format = *audio_format;
diff --git a/src/encoder/wave_encoder.c b/src/encoder/wave_encoder.c
index e66cc1917..f34ae0241 100644
--- a/src/encoder/wave_encoder.c
+++ b/src/encoder/wave_encoder.c
@@ -114,16 +114,39 @@ wave_encoder_open(struct encoder *_encoder,
struct wave_encoder *encoder = (struct wave_encoder *)_encoder;
void *buffer;
- encoder->bits = audio_format->bits;
+ assert(audio_format_valid(audio_format));
+
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S8:
+ encoder->bits = 8;
+ break;
+
+ case SAMPLE_FORMAT_S16:
+ encoder->bits = 16;
+ break;
+
+ case SAMPLE_FORMAT_S24_P32:
+ encoder->bits = 24;
+ break;
+
+ case SAMPLE_FORMAT_S32:
+ encoder->bits = 32;
+ break;
+
+ default:
+ audio_format->format = SAMPLE_FORMAT_S16;
+ encoder->bits = 16;
+ break;
+ }
buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) );
/* create PCM wave header in initial buffer */
fill_wave_header((struct wave_header *) buffer,
audio_format->channels,
- audio_format->bits,
+ encoder->bits,
audio_format->sample_rate,
- (audio_format->bits / 8) * audio_format->channels );
+ (encoder->bits / 8) * audio_format->channels );
encoder->buffer_length = sizeof(struct wave_header);
return true;
diff --git a/src/filter/volume_filter_plugin.c b/src/filter/volume_filter_plugin.c
index f6639a75f..285a4b7a4 100644
--- a/src/filter/volume_filter_plugin.c
+++ b/src/filter/volume_filter_plugin.c
@@ -75,8 +75,9 @@ volume_filter_open(struct filter *_filter,
{
struct volume_filter *filter = (struct volume_filter *)_filter;
- if (audio_format->bits != 8 && audio_format->bits != 16 &&
- audio_format->bits != 24) {
+ if (audio_format->format != SAMPLE_FORMAT_S8 &&
+ audio_format->format != SAMPLE_FORMAT_S16 &&
+ audio_format->format != SAMPLE_FORMAT_S24_P32) {
g_set_error(error_r, volume_quark(), 0,
"Unsupported audio format");
return false;
diff --git a/src/normalize.c b/src/normalize.c
index f9201df64..1c8173def 100644
--- a/src/normalize.c
+++ b/src/normalize.c
@@ -47,7 +47,8 @@ void finishNormalization(void)
void normalizeData(void *buffer, int bufferSize,
const struct audio_format *format)
{
- if ((format->bits != 16) || (format->channels != 2)) return;
+ if (format->format != SAMPLE_FORMAT_S16 || format->channels != 2)
+ return;
Compressor_Process_int16(compressor, buffer, bufferSize / 2);
}
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 2c642015d..b7325de07 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -185,13 +185,22 @@ alsa_test_default_device(void)
static snd_pcm_format_t
get_bitformat(const struct audio_format *af)
{
- switch (af->bits) {
- case 8: return SND_PCM_FORMAT_S8;
- case 16: return SND_PCM_FORMAT_S16;
- case 24: return SND_PCM_FORMAT_S24;
- case 32: return SND_PCM_FORMAT_S32;
+ switch (af->format) {
+ case SAMPLE_FORMAT_S8:
+ return SND_PCM_FORMAT_S8;
+
+ case SAMPLE_FORMAT_S16:
+ return SND_PCM_FORMAT_S16;
+
+ case SAMPLE_FORMAT_S24_P32:
+ return SND_PCM_FORMAT_S24;
+
+ case SAMPLE_FORMAT_S32:
+ return SND_PCM_FORMAT_S32;
+
+ default:
+ return SND_PCM_FORMAT_UNKNOWN;
}
- return SND_PCM_FORMAT_UNKNOWN;
}
static snd_pcm_format_t
@@ -264,61 +273,67 @@ configure_hw:
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
byteswap_bitformat(bitformat));
if (err == 0) {
- g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n",
- alsa_device(ad), audio_format->bits);
+ g_debug("ALSA device \"%s\": converting format %s to reverse-endian",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
audio_format->reverse_endian = 1;
}
}
- if (err == -EINVAL && (audio_format->bits == 24 ||
- audio_format->bits == 16)) {
+ if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 ||
+ audio_format->format == SAMPLE_FORMAT_S16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S32);
if (err == 0) {
- g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n",
- alsa_device(ad), audio_format->bits);
- audio_format->bits = 32;
+ g_debug("ALSA device \"%s\": converting format %s to 32 bit\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S32;
}
}
- if (err == -EINVAL && (audio_format->bits == 24 ||
- audio_format->bits == 16)) {
+ if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 ||
+ audio_format->format == SAMPLE_FORMAT_S16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
byteswap_bitformat(SND_PCM_FORMAT_S32));
if (err == 0) {
- g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n",
- alsa_device(ad), audio_format->bits);
- audio_format->bits = 32;
+ g_debug("ALSA device \"%s\": converting format %s to 32 bit backward-endian\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S32;
audio_format->reverse_endian = 1;
}
}
- if (err == -EINVAL && audio_format->bits != 16) {
+ if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) {
/* fall back to 16 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S16);
if (err == 0) {
- g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n",
- alsa_device(ad), audio_format->bits);
- audio_format->bits = 16;
+ g_debug("ALSA device \"%s\": converting format %s to 16 bit\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S16;
}
}
- if (err == -EINVAL && audio_format->bits != 16) {
+ if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) {
/* fall back to 16 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
byteswap_bitformat(SND_PCM_FORMAT_S16));
if (err == 0) {
- g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n",
- alsa_device(ad), audio_format->bits);
- audio_format->bits = 16;
+ g_debug("ALSA device \"%s\": converting format %s to 16 bit backward-endian\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S16;
audio_format->reverse_endian = 1;
}
}
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support %u bit audio: %s",
- alsa_device(ad), audio_format->bits,
+ "ALSA device \"%s\" does not support format %s: %s",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format),
snd_strerror(-err));
return false;
}
@@ -449,7 +464,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error)
/* sample format is not supported by this plugin -
fall back to 16 bit samples */
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
bitformat = SND_PCM_FORMAT_S16;
}
diff --git a/src/output/ao_plugin.c b/src/output/ao_plugin.c
index d69175272..7afca0db2 100644
--- a/src/output/ao_plugin.c
+++ b/src/output/ao_plugin.c
@@ -170,13 +170,24 @@ ao_output_open(void *data, struct audio_format *audio_format,
ao_sample_format format;
struct ao_data *ad = (struct ao_data *)data;
- /* support for 24 bit samples in libao is currently dubious,
- and until we have sorted that out, resample everything to
- 16 bit */
- if (audio_format->bits > 16)
- audio_format->bits = 16;
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S8:
+ format.bits = 8;
+ break;
+
+ case SAMPLE_FORMAT_S16:
+ format.bits = 16;
+ break;
+
+ default:
+ /* support for 24 bit samples in libao is currently
+ dubious, and until we have sorted that out,
+ convert everything to 16 bit */
+ audio_format->format = SAMPLE_FORMAT_S16;
+ format.bits = 16;
+ break;
+ }
- format.bits = audio_format->bits;
format.rate = audio_format->sample_rate;
format.byte_format = AO_FMT_NATIVE;
format.channels = audio_format->channels;
diff --git a/src/output/jack_output_plugin.c b/src/output/jack_output_plugin.c
index 7e5a52993..f50bc37d0 100644
--- a/src/output/jack_output_plugin.c
+++ b/src/output/jack_output_plugin.c
@@ -157,8 +157,9 @@ set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
else if (audio_format->channels > jd->num_source_ports)
audio_format->channels = 2;
- if (audio_format->bits != 16 && audio_format->bits != 24)
- audio_format->bits = 24;
+ if (audio_format->format != SAMPLE_FORMAT_S16 &&
+ audio_format->format != SAMPLE_FORMAT_S24_P32)
+ audio_format->format = SAMPLE_FORMAT_S24_P32;
}
static void
@@ -606,13 +607,13 @@ static void
mpd_jack_write_samples(struct jack_data *jd, const void *src,
unsigned num_samples)
{
- switch (jd->audio_format.bits) {
- case 16:
+ switch (jd->audio_format.format) {
+ case SAMPLE_FORMAT_S16:
mpd_jack_write_samples_16(jd, (const int16_t*)src,
num_samples);
break;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
mpd_jack_write_samples_24(jd, (const int32_t*)src,
num_samples);
break;
diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c
index 5a9a9b48b..86f147e5a 100644
--- a/src/output/mvp_plugin.c
+++ b/src/output/mvp_plugin.c
@@ -172,19 +172,19 @@ mvp_set_pcm_params(struct mvp_data *md, struct audio_format *audio_format,
}
/* 0,1=24bit(24) , 2,3=16bit */
- switch (audio_format->bits) {
- case 16:
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S16:
mix[1] = 2;
break;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
mix[1] = 0;
break;
default:
- g_debug("unsupported sample format %u - falling back to stereo",
- audio_format->bits);
- audio_format->bits = 16;
+ g_debug("unsupported sample format %s - falling back to 16 bit",
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S16;
mix[1] = 2;
break;
}
diff --git a/src/output/openal_plugin.c b/src/output/openal_plugin.c
index 8fda110e1..0aded4d9a 100644
--- a/src/output/openal_plugin.c
+++ b/src/output/openal_plugin.c
@@ -58,25 +58,29 @@ openal_output_quark(void)
static ALenum
openal_audio_format(struct audio_format *audio_format)
{
- /* Only 8 and 16 bit samples are supported */
- if (audio_format->bits != 16 && audio_format->bits != 8)
- audio_format->bits = 16;
-
- switch (audio_format->bits)
- {
- case 16:
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S16:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO16;
if (audio_format->channels == 1)
return AL_FORMAT_MONO16;
break;
- case 8:
+ case SAMPLE_FORMAT_S8:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO8;
if (audio_format->channels == 1)
return AL_FORMAT_MONO8;
break;
+
+ default:
+ /* fall back to 16 bit */
+ audio_format->format = SAMPLE_FORMAT_S16;
+ if (audio_format->channels == 2)
+ return AL_FORMAT_STEREO16;
+ if (audio_format->channels == 1)
+ return AL_FORMAT_MONO16;
+ break;
}
return 0;
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
index b02d7d62e..f16374e39 100644
--- a/src/output/oss_plugin.c
+++ b/src/output/oss_plugin.c
@@ -490,17 +490,18 @@ oss_setup(struct oss_data *od, GError **error)
}
od->audio_format.sample_rate = tmp;
- switch (od->audio_format.bits) {
- case 8:
+ switch (od->audio_format.format) {
+ case SAMPLE_FORMAT_S8:
tmp = AFMT_S8;
break;
- case 16:
+
+ case SAMPLE_FORMAT_S16:
tmp = AFMT_S16_MPD;
break;
default:
/* not supported by OSS - fall back to 16 bit */
- od->audio_format.bits = 16;
+ od->audio_format.format = SAMPLE_FORMAT_S16;
tmp = AFMT_S16_MPD;
break;
}
diff --git a/src/output/osx_plugin.c b/src/output/osx_plugin.c
index afcd143b3..22b742ee5 100644
--- a/src/output/osx_plugin.c
+++ b/src/output/osx_plugin.c
@@ -166,9 +166,6 @@ osx_output_open(void *data, struct audio_format *audio_format, GError **error)
OSStatus status;
ComponentResult result;
- if (audio_format->bits > 16)
- audio_format->bits = 16;
-
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
@@ -226,7 +223,21 @@ osx_output_open(void *data, struct audio_format *audio_format, GError **error)
stream_description.mFramesPerPacket = 1;
stream_description.mBytesPerFrame = stream_description.mBytesPerPacket;
stream_description.mChannelsPerFrame = audio_format->channels;
- stream_description.mBitsPerChannel = audio_format->bits;
+
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S8:
+ stream_description.mBitsPerChannel = 8;
+ break;
+
+ case SAMPLE_FORMAT_S16:
+ stream_description.mBitsPerChannel = 16;
+ break;
+
+ default:
+ audio_format->format = SAMPLE_FORMAT_S16;
+ stream_description.mBitsPerChannel = 16;
+ break;
+ }
result = AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0,
diff --git a/src/output/pulse_output_plugin.c b/src/output/pulse_output_plugin.c
index 3da1b3593..a64157920 100644
--- a/src/output/pulse_output_plugin.c
+++ b/src/output/pulse_output_plugin.c
@@ -467,7 +467,7 @@ pulse_output_open(void *data, struct audio_format *audio_format,
/* MPD doesn't support the other pulseaudio sample formats, so
we just force MPD to send us everything as 16 bit */
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
ss.format = PA_SAMPLE_S16NE;
ss.rate = audio_format->sample_rate;
diff --git a/src/output/solaris_output_plugin.c b/src/output/solaris_output_plugin.c
index b187630ee..fe84068f1 100644
--- a/src/output/solaris_output_plugin.c
+++ b/src/output/solaris_output_plugin.c
@@ -89,7 +89,7 @@ solaris_output_open(void *data, struct audio_format *audio_format,
/* support only 16 bit mono/stereo for now; nothing else has
been tested */
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
/* open the device in non-blocking mode */
@@ -119,7 +119,7 @@ solaris_output_open(void *data, struct audio_format *audio_format,
info.play.sample_rate = audio_format->sample_rate;
info.play.channels = audio_format->channels;
- info.play.precision = audio_format->bits;
+ info.play.precision = 16;
info.play.encoding = AUDIO_ENCODING_LINEAR;
ret = ioctl(so->fd, AUDIO_SETINFO, &info);
diff --git a/src/pcm_convert.c b/src/pcm_convert.c
index 1d6299964..8d529dd5f 100644
--- a/src/pcm_convert.c
+++ b/src/pcm_convert.c
@@ -63,15 +63,15 @@ pcm_convert_16(struct pcm_convert_state *state,
const int16_t *buf;
size_t len;
- assert(dest_format->bits == 16);
+ assert(dest_format->format == SAMPLE_FORMAT_S16);
buf = pcm_convert_to_16(&state->format_buffer, &state->dither,
- src_format->bits, src_buffer, src_size,
+ src_format->format, src_buffer, src_size,
&len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
- "Conversion from %u to 16 bit is not implemented",
- src_format->bits);
+ "Conversion from %s to 16 bit is not implemented",
+ sample_format_to_string(src_format->format));
return NULL;
}
@@ -119,14 +119,14 @@ pcm_convert_24(struct pcm_convert_state *state,
const int32_t *buf;
size_t len;
- assert(dest_format->bits == 24);
+ assert(dest_format->format == SAMPLE_FORMAT_S24_P32);
- buf = pcm_convert_to_24(&state->format_buffer, src_format->bits,
+ buf = pcm_convert_to_24(&state->format_buffer, src_format->format,
src_buffer, src_size, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
- "Conversion from %u to 24 bit is not implemented",
- src_format->bits);
+ "Conversion from %s to 24 bit is not implemented",
+ sample_format_to_string(src_format->format));
return NULL;
}
@@ -174,14 +174,14 @@ pcm_convert_32(struct pcm_convert_state *state,
const int32_t *buf;
size_t len;
- assert(dest_format->bits == 32);
+ assert(dest_format->format == SAMPLE_FORMAT_S32);
- buf = pcm_convert_to_32(&state->format_buffer, src_format->bits,
+ buf = pcm_convert_to_32(&state->format_buffer, src_format->format,
src_buffer, src_size, &len);
if (buf == NULL) {
g_set_error(error_r, pcm_convert_quark(), 0,
- "Conversion from %u to 24 bit is not implemented",
- src_format->bits);
+ "Conversion from %s to 24 bit is not implemented",
+ sample_format_to_string(src_format->format));
return NULL;
}
@@ -227,20 +227,20 @@ pcm_convert(struct pcm_convert_state *state,
size_t *dest_size_r,
GError **error_r)
{
- switch (dest_format->bits) {
- case 16:
+ switch (dest_format->format) {
+ case SAMPLE_FORMAT_S16:
return pcm_convert_16(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
return pcm_convert_24(state,
src_format, src, src_size,
dest_format, dest_size_r,
error_r);
- case 32:
+ case SAMPLE_FORMAT_S32:
return pcm_convert_32(state,
src_format, src, src_size,
dest_format, dest_size_r,
@@ -248,8 +248,8 @@ pcm_convert(struct pcm_convert_state *state,
default:
g_set_error(error_r, pcm_convert_quark(), 0,
- "PCM conversion to %u bit is not implemented",
- dest_format->bits);
+ "PCM conversion to %s is not implemented",
+ sample_format_to_string(dest_format->format));
return NULL;
}
}
diff --git a/src/pcm_format.c b/src/pcm_format.c
index 8da253db9..b0dad2ba3 100644
--- a/src/pcm_format.c
+++ b/src/pcm_format.c
@@ -50,14 +50,17 @@ pcm_convert_32_to_16(struct pcm_dither *dither,
const int16_t *
pcm_convert_to_16(struct pcm_buffer *buffer, struct pcm_dither *dither,
- uint8_t bits, const void *src,
+ enum sample_format src_format, const void *src,
size_t src_size, size_t *dest_size_r)
{
unsigned num_samples;
int16_t *dest;
- switch (bits) {
- case 8:
+ switch (src_format) {
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+
+ case SAMPLE_FORMAT_S8:
num_samples = src_size;
*dest_size_r = src_size * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -67,11 +70,11 @@ pcm_convert_to_16(struct pcm_buffer *buffer, struct pcm_dither *dither,
num_samples);
return dest;
- case 16:
+ case SAMPLE_FORMAT_S16:
*dest_size_r = src_size;
return src;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
num_samples = src_size / 4;
*dest_size_r = num_samples * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -81,7 +84,7 @@ pcm_convert_to_16(struct pcm_buffer *buffer, struct pcm_dither *dither,
num_samples);
return dest;
- case 32:
+ case SAMPLE_FORMAT_S32:
num_samples = src_size / 4;
*dest_size_r = num_samples * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -127,14 +130,17 @@ pcm_convert_32_to_24(int32_t *out, const int16_t *in,
const int32_t *
pcm_convert_to_24(struct pcm_buffer *buffer,
- uint8_t bits, const void *src,
+ enum sample_format src_format, const void *src,
size_t src_size, size_t *dest_size_r)
{
unsigned num_samples;
int32_t *dest;
- switch (bits) {
- case 8:
+ switch (src_format) {
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+
+ case SAMPLE_FORMAT_S8:
num_samples = src_size;
*dest_size_r = src_size * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -143,7 +149,7 @@ pcm_convert_to_24(struct pcm_buffer *buffer,
num_samples);
return dest;
- case 16:
+ case SAMPLE_FORMAT_S16:
num_samples = src_size / 2;
*dest_size_r = num_samples * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -152,11 +158,11 @@ pcm_convert_to_24(struct pcm_buffer *buffer,
num_samples);
return dest;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
*dest_size_r = src_size;
return src;
- case 32:
+ case SAMPLE_FORMAT_S32:
num_samples = src_size / 4;
*dest_size_r = num_samples * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -201,14 +207,17 @@ pcm_convert_24_to_32(int32_t *out, const int32_t *in,
const int32_t *
pcm_convert_to_32(struct pcm_buffer *buffer,
- uint8_t bits, const void *src,
+ enum sample_format src_format, const void *src,
size_t src_size, size_t *dest_size_r)
{
unsigned num_samples;
int32_t *dest;
- switch (bits) {
- case 8:
+ switch (src_format) {
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+
+ case SAMPLE_FORMAT_S8:
num_samples = src_size;
*dest_size_r = src_size * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -217,7 +226,7 @@ pcm_convert_to_32(struct pcm_buffer *buffer,
num_samples);
return dest;
- case 16:
+ case SAMPLE_FORMAT_S16:
num_samples = src_size / 2;
*dest_size_r = num_samples * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -226,7 +235,7 @@ pcm_convert_to_32(struct pcm_buffer *buffer,
num_samples);
return dest;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
num_samples = src_size / 4;
*dest_size_r = num_samples * sizeof(*dest);
dest = pcm_buffer_get(buffer, *dest_size_r);
@@ -235,7 +244,7 @@ pcm_convert_to_32(struct pcm_buffer *buffer,
num_samples);
return dest;
- case 32:
+ case SAMPLE_FORMAT_S32:
*dest_size_r = src_size;
return src;
}
diff --git a/src/pcm_format.h b/src/pcm_format.h
index 350566827..6ea5573bd 100644
--- a/src/pcm_format.h
+++ b/src/pcm_format.h
@@ -20,6 +20,8 @@
#ifndef PCM_FORMAT_H
#define PCM_FORMAT_H
+#include "audio_format.h"
+
#include <stdint.h>
#include <stddef.h>
@@ -40,7 +42,7 @@ struct pcm_dither;
*/
const int16_t *
pcm_convert_to_16(struct pcm_buffer *buffer, struct pcm_dither *dither,
- uint8_t bits, const void *src,
+ enum sample_format src_format, const void *src,
size_t src_size, size_t *dest_size_r);
/**
@@ -55,7 +57,7 @@ pcm_convert_to_16(struct pcm_buffer *buffer, struct pcm_dither *dither,
*/
const int32_t *
pcm_convert_to_24(struct pcm_buffer *buffer,
- uint8_t bits, const void *src,
+ enum sample_format src_format, const void *src,
size_t src_size, size_t *dest_size_r);
/**
@@ -70,7 +72,7 @@ pcm_convert_to_24(struct pcm_buffer *buffer,
*/
const int32_t *
pcm_convert_to_32(struct pcm_buffer *buffer,
- uint8_t bits, const void *src,
+ enum sample_format src_format, const void *src,
size_t src_size, size_t *dest_size_r);
#endif
diff --git a/src/pcm_mix.c b/src/pcm_mix.c
index 6e678a912..0f767a1d9 100644
--- a/src/pcm_mix.c
+++ b/src/pcm_mix.c
@@ -103,18 +103,18 @@ pcm_add(void *buffer1, const void *buffer2, size_t size,
int vol1, int vol2,
const struct audio_format *format)
{
- switch (format->bits) {
- case 8:
+ switch (format->format) {
+ case SAMPLE_FORMAT_S8:
pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2,
size, vol1, vol2);
break;
- case 16:
+ case SAMPLE_FORMAT_S16:
pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2,
size / 2, vol1, vol2);
break;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
pcm_add_24((int32_t*)buffer1,
(const int32_t*)buffer2,
size / 4, vol1, vol2);
@@ -127,7 +127,8 @@ pcm_add(void *buffer1, const void *buffer2, size_t size,
break;
default:
- g_error("%u bits not supported by pcm_add!\n", format->bits);
+ g_error("format %s not supported by pcm_add",
+ sample_format_to_string(format->format));
}
}
diff --git a/src/pcm_volume.c b/src/pcm_volume.c
index 90ad17d6d..acb6c41a8 100644
--- a/src/pcm_volume.c
+++ b/src/pcm_volume.c
@@ -150,17 +150,17 @@ pcm_volume(void *buffer, int length,
return true;
}
- switch (format->bits) {
- case 8:
+ switch (format->format) {
+ case SAMPLE_FORMAT_S8:
pcm_volume_change_8((int8_t *)buffer, length, volume);
return true;
- case 16:
+ case SAMPLE_FORMAT_S16:
pcm_volume_change_16((int16_t *)buffer, length / 2,
volume);
return true;
- case 24:
+ case SAMPLE_FORMAT_S24_P32:
pcm_volume_change_24((int32_t*)buffer, length / 4,
volume);
return true;
--
cgit v1.2.3