From 37754559b8f934ce8d554e0d9f976d4f6eb376d9 Mon Sep 17 00:00:00 2001
From: David Woodhouse <David.Woodhouse@intel.com>
Date: Sun, 19 Jul 2009 16:24:43 +0100
Subject: Add audio_format_init() function
It makes no difference right now, but we're about to add an endianness
flag and will want to make sure it's correctly initialised every time.
---
src/decoder/_flac_common.c | 5 ++---
src/decoder/audiofile_plugin.c | 8 +++-----
src/decoder/faad_plugin.c | 6 +-----
src/decoder/ffmpeg_plugin.c | 9 +++++----
src/decoder/mad_plugin.c | 10 ++--------
src/decoder/mikmod_plugin.c | 4 +---
src/decoder/modplug_plugin.c | 4 +---
src/decoder/mp4ff_plugin.c | 6 +-----
src/decoder/mpcdec_plugin.c | 4 +---
src/decoder/sidplay_plugin.cxx | 4 +---
src/decoder/sndfile_decoder_plugin.c | 4 +---
src/decoder/vorbis_plugin.c | 3 +--
src/decoder/wavpack_plugin.c | 6 +++---
13 files changed, 23 insertions(+), 50 deletions(-)
(limited to 'src/decoder')
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index 713dfe9b2..7b3453854 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -195,9 +195,8 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- data->audio_format.bits = (int8_t)si->bits_per_sample;
- data->audio_format.sample_rate = si->sample_rate;
- data->audio_format.channels = (int8_t)si->channels;
+ audio_format_init(&data->audio_format, si->sample_rate,
+ si->bits_per_sample, si->channels);
data->total_time = ((float)si->total_samples) / (si->sample_rate);
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index f66d90dc1..b4959f6c2 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -136,11 +136,9 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, bits);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- audio_format.bits = (uint8_t)bits;
- audio_format.sample_rate =
- (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
- audio_format.channels =
- (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+
+ audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK),
+ bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK));
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index d0537dd5b..1b8b2b784 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -262,11 +262,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
decoder_buffer_consume(buffer, nbytes);
- *audio_format = (struct audio_format){
- .bits = 16,
- .channels = channels,
- .sample_rate = sample_rate,
- };
+ audio_format_init(audio_format, sample_rate, 16, channels);
return true;
}
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index 03c46a732..f6003d2f3 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -267,6 +267,7 @@ 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;
@@ -275,13 +276,13 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
}
#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
- audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+ bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
#else
/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
- audio_format.bits = (uint8_t) 16;
+ bits = (uint8_t) 16;
#endif
- audio_format.sample_rate = (unsigned int)codec_context->sample_rate;
- audio_format.channels = codec_context->channels;
+ audio_format_init(&audio_format, codec_context->sample_rate, bits,
+ codec_context->channels);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index c6b9d32d3..85f4506d2 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -1148,13 +1148,6 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
return ret != DECODE_BREAK;
}
-static void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
-{
- af->bits = 24;
- af->sample_rate = (data->frame).header.samplerate;
- af->channels = MAD_NCHANNELS(&(data->frame).header);
-}
-
static void
mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
{
@@ -1170,7 +1163,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
return;
}
- mp3_audio_format(&data, &audio_format);
+ audio_format_init(&audio_format, data.frame.header.samplerate, 24,
+ MAD_NCHANNELS(&data.frame.header));
decoder_initialized(decoder, &audio_format,
data.input_stream->seekable, data.total_time);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 065c34319..e7b7bfb03 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -175,9 +175,7 @@ mod_decode(struct decoder *decoder, const char *path)
return;
}
- audio_format.bits = 16;
- audio_format.sample_rate = 44100;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 44100, 16, 2);
secPerByte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index 31f0a47c2..6c375e6a0 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -121,9 +121,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
return;
}
- audio_format.bits = 16;
- audio_format.sample_rate = 44100;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 44100, 16, 2);
sec_perbyte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index cf9382904..d2c63f983 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -131,11 +131,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
}
*track_r = track;
- *audio_format = (struct audio_format){
- .bits = 16,
- .channels = channels,
- .sample_rate = sample_rate,
- };
+ audio_format_init(audio_format, sample_rate, 16, channels);
if (!audio_format_valid(audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index 26349f93a..a684da104 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -193,9 +193,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
mpc_demux_get_info(demux, &info);
#endif
- audio_format.bits = 24;
- audio_format.channels = info.channels;
- audio_format.sample_rate = info.sample_freq;
+ audio_format_init(&audio_format, info.sample_freq, 24, info.channels);
if (!audio_format_valid(&audio_format)) {
#ifndef MPC_IS_OLD_API
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index c62e6b4b6..54ab746e2 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -103,9 +103,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
/* initialize the MPD decoder */
struct audio_format audio_format;
- audio_format.sample_rate = 48000;
- audio_format.bits = 16;
- audio_format.channels = 2;
+ audio_format_init(&audio_format, 48000, 16, 2);
decoder_initialized(decoder, &audio_format, false, -1);
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
index 0c5d2f063..4cc64459f 100644
--- a/src/decoder/sndfile_decoder_plugin.c
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -124,12 +124,10 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
return;
}
- audio_format.sample_rate = info.samplerate;
/* 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. */
- audio_format.bits = 32;
- audio_format.channels = info.channels;
+ audio_format_init(&audio_format, info.samplerate, 32, info.channels);
if (!audio_format_valid(&audio_format)) {
g_warning("invalid audio format");
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index d4f81e91f..bab1d57ec 100644
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -324,8 +324,7 @@ vorbis_stream_decode(struct decoder *decoder,
vorbis_info *vi = ov_info(&vf, -1);
struct replay_gain_info *new_rgi;
- audio_format.channels = vi->channels;
- audio_format.sample_rate = vi->rate;
+ audio_format_init(&audio_format, vi->rate, 16, vi->channels);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 821536fb5..f3d701144 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -145,9 +145,9 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
int bytes_per_sample, output_sample_size;
int position;
- audio_format.sample_rate = WavpackGetSampleRate(wpc);
- audio_format.channels = WavpackGetReducedChannels(wpc);
- audio_format.bits = WavpackGetBitsPerSample(wpc);
+ audio_format_init(&audio_format, WavpackGetSampleRate(wpc),
+ WavpackGetBitsPerSample(wpc),
+ WavpackGetReducedChannels(wpc));
/* round bitwidth to 8-bit units */
audio_format.bits = (audio_format.bits + 7) & (~7);
--
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