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-rw-r--r--src/inputPlugins/_flac_common.c7
-rw-r--r--src/inputPlugins/_flac_common.h1
-rw-r--r--src/inputPlugins/aac_plugin.c11
-rw-r--r--src/inputPlugins/audiofile_plugin.c21
-rw-r--r--src/inputPlugins/flac_plugin.c10
-rw-r--r--src/inputPlugins/mod_plugin.c15
-rw-r--r--src/inputPlugins/mp3_plugin.c6
-rw-r--r--src/inputPlugins/mp4_plugin.c15
-rw-r--r--src/inputPlugins/mpc_plugin.c21
-rw-r--r--src/inputPlugins/oggflac_plugin.c12
-rw-r--r--src/inputPlugins/oggvorbis_plugin.c13
-rw-r--r--src/inputPlugins/wavpack_plugin.c27
12 files changed, 78 insertions, 81 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index e1017e596..c48f43da9 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -162,11 +162,10 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- dc.audioFormat.bits = (mpd_sint8)si->bits_per_sample;
- dc.audioFormat.sampleRate = si->sample_rate;
- dc.audioFormat.channels = (mpd_sint8)si->channels;
+ data->audio_format.bits = (mpd_sint8)si->bits_per_sample;
+ data->audio_format.sampleRate = si->sample_rate;
+ data->audio_format.channels = (mpd_sint8)si->channels;
dc.totalTime = ((float)si->total_samples) / (si->sample_rate);
- getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
flacParseReplayGain(block, data);
diff --git a/src/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h
index 9eb916b8c..e87ae9307 100644
--- a/src/inputPlugins/_flac_common.h
+++ b/src/inputPlugins/_flac_common.h
@@ -143,6 +143,7 @@ typedef struct {
size_t chunk_length;
float time;
unsigned int bitRate;
+ AudioFormat audio_format;
FLAC__uint64 position;
struct decoder *decoder;
InputStream *inStream;
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index 8765dc21c..4fa54c646 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -286,6 +286,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
long bread;
+ AudioFormat audio_format;
uint32_t sampleRate;
unsigned char channels;
int eof = 0;
@@ -335,7 +336,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
return -1;
}
- dc.audioFormat.bits = 16;
+ audio_format.bits = 16;
dc.totalTime = totalTime;
@@ -369,11 +370,9 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
#endif
if (dc.state != DECODE_STATE_DECODE) {
- dc.audioFormat.channels = frameInfo.channels;
- dc.audioFormat.sampleRate = sampleRate;
- getOutputAudioFormat(&(dc.audioFormat),
- &(ob.audioFormat));
- decoder_initialized(mpd_decoder);
+ audio_format.channels = frameInfo.channels;
+ audio_format.sampleRate = sampleRate;
+ decoder_initialized(mpd_decoder, &audio_format);
}
advanceAacBuffer(&b, frameInfo.bytesconsumed);
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 696621aff..f96ea2fca 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -45,6 +45,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
int fs, frame_count;
AFfilehandle af_fp;
int bits;
+ AudioFormat audio_format;
mpd_uint16 bitRate;
struct stat st;
@@ -62,30 +63,30 @@ static int audiofile_decode(struct decoder * decoder, char *path)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- dc.audioFormat.bits = (mpd_uint8)bits;
- dc.audioFormat.sampleRate =
+ audio_format.bits = (mpd_uint8)bits;
+ audio_format.sampleRate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
- dc.audioFormat.channels =
+ audio_format.channels =
(mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
- getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
dc.totalTime =
- ((float)frame_count / (float)dc.audioFormat.sampleRate);
+ ((float)frame_count / (float)audio_format.sampleRate);
bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.totalTime / 1000.0 + 0.5);
- if (dc.audioFormat.bits != 8 && dc.audioFormat.bits != 16) {
+ if (audio_format.bits != 8 && audio_format.bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
- path, dc.audioFormat.bits);
+ path, audio_format.bits);
afCloseFile(af_fp);
return -1;
}
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
- decoder_initialized(decoder);
+ decoder_initialized(decoder, &audio_format);
+
{
int ret, eof = 0, current = 0;
char chunk[CHUNK_SIZE];
@@ -94,7 +95,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
if (dc.command == DECODE_COMMAND_SEEK) {
decoder_clear(decoder);
current = dc.seekWhere *
- dc.audioFormat.sampleRate;
+ audio_format.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
dc_command_finished();
}
@@ -110,7 +111,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
1,
chunk, ret * fs,
(float)current /
- (float)dc.audioFormat.
+ (float)audio_format.
sampleRate, bitRate,
NULL);
if (dc.command == DECODE_COMMAND_STOP)
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index 4bfc87246..f5d5469f7 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -242,7 +242,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
FLAC__uint32 samples = frame->header.blocksize;
unsigned int c_samp;
const unsigned int num_channels = frame->header.channels;
- const unsigned int bytes_per_sample = (dc.audioFormat.bits / 8);
+ const unsigned int bytes_per_sample = (data->audio_format.bits / 8);
const unsigned int bytes_per_channel =
bytes_per_sample * frame->header.channels;
const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
@@ -250,7 +250,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
float timeChange;
FLAC__uint64 newPosition = 0;
- assert(dc.audioFormat.bits > 0);
+ assert(data->audio_format.bits > 0);
timeChange = ((float)samples) / frame->header.sample_rate;
data->time += timeChange;
@@ -415,7 +415,7 @@ static int flac_decode_internal(struct decoder * decoder,
}
}
- decoder_initialized(decoder);
+ decoder_initialized(decoder, &data.audio_format);
while (1) {
if (!flac_process_single(flacDec))
@@ -424,11 +424,11 @@ static int flac_decode_internal(struct decoder * decoder,
break;
if (dc.command == DECODE_COMMAND_SEEK) {
FLAC__uint64 sampleToSeek = dc.seekWhere *
- dc.audioFormat.sampleRate + 0.5;
+ data.audio_format.sampleRate + 0.5;
if (flac_seek_absolute(flacDec, sampleToSeek)) {
decoder_clear(decoder);
data.time = ((float)sampleToSeek) /
- dc.audioFormat.sampleRate;
+ data.audio_format.sampleRate;
data.position = 0;
} else
dc.seekError = 1;
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 930b041e9..4ab1338bf 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -162,6 +162,7 @@ static void mod_close(mod_Data * data)
static int mod_decode(struct decoder * decoder, char *path)
{
mod_Data *data;
+ AudioFormat audio_format;
float total_time = 0.0;
int ret;
float secPerByte;
@@ -176,16 +177,16 @@ static int mod_decode(struct decoder * decoder, char *path)
}
dc.totalTime = 0;
- dc.audioFormat.bits = 16;
- dc.audioFormat.sampleRate = 44100;
- dc.audioFormat.channels = 2;
- getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
+ audio_format.bits = 16;
+ audio_format.sampleRate = 44100;
+ audio_format.channels = 2;
secPerByte =
- 1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) *
- (float)dc.audioFormat.sampleRate);
+ 1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
+ (float)audio_format.sampleRate);
+
+ decoder_initialized(decoder, &audio_format);
- decoder_initialized(decoder);
while (1) {
if (dc.command == DECODE_COMMAND_SEEK) {
dc.seekError = 1;
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 9959f8601..a7f39a3a4 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -1021,6 +1021,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
mp3DecodeData data;
MpdTag *tag = NULL;
ReplayGainInfo *replayGainInfo = NULL;
+ AudioFormat audio_format;
if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) <
0) {
@@ -1032,8 +1033,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
return 0;
}
- initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat));
- getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
+ initAudioFormatFromMp3DecodeData(&data, &audio_format);
dc.totalTime = data.totalTime;
@@ -1062,7 +1062,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
freeMpdTag(tag);
}
- decoder_initialized(decoder);
+ decoder_initialized(decoder, &audio_format);
while (mp3Read(&data, decoder, &replayGainInfo) != DECODE_BREAK) ;
/* send last little bit if not DECODE_COMMAND_STOP */
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index f917ec345..d2c0f1b6c 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -88,6 +88,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
+ AudioFormat audio_format;
unsigned char *mp4Buffer;
unsigned int mp4BufferSize;
uint32_t sampleRate;
@@ -139,7 +140,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
#endif
faacDecSetConfiguration(decoder, config);
- dc.audioFormat.bits = 16;
+ audio_format.bits = 16;
mp4Buffer = NULL;
mp4BufferSize = 0;
@@ -154,8 +155,8 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
return -1;
}
- dc.audioFormat.sampleRate = sampleRate;
- dc.audioFormat.channels = channels;
+ audio_format.sampleRate = sampleRate;
+ audio_format.channels = channels;
file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
@@ -245,11 +246,9 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
#endif
- dc.audioFormat.sampleRate = scale;
- dc.audioFormat.channels = frameInfo.channels;
- getOutputAudioFormat(&(dc.audioFormat),
- &(ob.audioFormat));
- decoder_initialized(mpd_decoder);
+ audio_format.sampleRate = scale;
+ audio_format.channels = frameInfo.channels;
+ decoder_initialized(mpd_decoder, &audio_format);
}
if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index 13f2adcee..5a57ea4c8 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -111,6 +111,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
mpc_decoder decoder;
mpc_reader reader;
mpc_streaminfo info;
+ AudioFormat audio_format;
MpcCallbackData data;
@@ -161,11 +162,9 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
dc.totalTime = mpc_streaminfo_get_length(&info);
- dc.audioFormat.bits = 16;
- dc.audioFormat.channels = info.channels;
- dc.audioFormat.sampleRate = info.sample_freq;
-
- getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
+ audio_format.bits = 16;
+ audio_format.channels = info.channels;
+ audio_format.sampleRate = info.sample_freq;
replayGainInfo = newReplayGainInfo();
replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -173,11 +172,11 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
replayGainInfo->trackGain = info.gain_title * 0.01;
replayGainInfo->trackPeak = info.peak_title / 32767.0;
- decoder_initialized(mpd_decoder);
+ decoder_initialized(mpd_decoder, &audio_format);
while (!eof) {
if (dc.command == DECODE_COMMAND_SEEK) {
- samplePos = dc.seekWhere * dc.audioFormat.sampleRate;
+ samplePos = dc.seekWhere * audio_format.sampleRate;
if (mpc_decoder_seek_sample(&decoder, samplePos)) {
decoder_clear(mpd_decoder);
s16 = (mpd_sint16 *) chunk;
@@ -210,10 +209,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
if (chunkpos >= MPC_CHUNK_SIZE) {
total_time = ((float)samplePos) /
- dc.audioFormat.sampleRate;
+ audio_format.sampleRate;
bitRate = vbrUpdateBits *
- dc.audioFormat.sampleRate / 1152 / 1000;
+ audio_format.sampleRate / 1152 / 1000;
decoder_data(mpd_decoder, inStream,
inStream->seekable,
@@ -232,10 +231,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
}
if (dc.command != DECODE_COMMAND_STOP && chunkpos > 0) {
- total_time = ((float)samplePos) / dc.audioFormat.sampleRate;
+ total_time = ((float)samplePos) / audio_format.sampleRate;
bitRate =
- vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000;
+ vbrUpdateBits * audio_format.sampleRate / 1152 / 1000;
decoder_data(mpd_decoder, NULL, inStream->seekable,
chunk, chunkpos, total_time, bitRate,
diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
index b5e73e455..5879b7054 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -188,7 +188,7 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(const
c_chan++) {
u16 = buf[c_chan][c_samp];
uc = (unsigned char *)&u16;
- for (i = 0; i < (dc.audioFormat.bits / 8); i++) {
+ for (i = 0; i < (data->audio_format.bits / 8); i++) {
if (data->chunk_length >= FLAC_CHUNK_SIZE) {
if (flacSendChunk(data) < 0) {
return
@@ -345,7 +345,7 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
goto fail;
}
- decoder_initialized(mpd_decoder);
+ decoder_initialized(mpd_decoder, &data.audio_format);
while (1) {
OggFLAC__seekable_stream_decoder_process_single(decoder);
@@ -353,14 +353,14 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
break;
}
- if (dc.command == DECODE_COMMAND_SEEK) {
- FLAC__uint64 sampleToSeek = dc.seekWhere *
- dc.audioFormat.sampleRate + 0.5;
+ if (dc->command == DECODE_COMMAND_SEEK) {
+ FLAC__uint64 sampleToSeek = dc->seekWhere *
+ data.audio_format.sampleRate + 0.5;
if (OggFLAC__seekable_stream_decoder_seek_absolute
(decoder, sampleToSeek)) {
decoder_clear(mpd_decoder);
data.time = ((float)sampleToSeek) /
- dc.audioFormat.sampleRate;
+ data.audio_format.sampleRate;
data.position = 0;
} else
dc.seekError = 1;
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index 9e1cf89c0..2fb0a9e52 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -215,6 +215,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
OggVorbis_File vf;
ov_callbacks callbacks;
OggCallbackData data;
+ AudioFormat audio_format;
int current_section;
int prev_section = -1;
long ret;
@@ -264,7 +265,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
dc.totalTime = ov_time_total(&vf, -1);
if (dc.totalTime < 0)
dc.totalTime = 0;
- dc.audioFormat.bits = 16;
+ audio_format.bits = 16;
while (1) {
if (dc.command == DECODE_COMMAND_SEEK) {
@@ -281,12 +282,10 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
if (current_section != prev_section) {
/*printf("new song!\n"); */
vorbis_info *vi = ov_info(&vf, -1);
- dc.audioFormat.channels = vi->channels;
- dc.audioFormat.sampleRate = vi->rate;
+ audio_format.channels = vi->channels;
+ audio_format.sampleRate = vi->rate;
if (dc.state == DECODE_STATE_START) {
- getOutputAudioFormat(&(dc.audioFormat),
- &(ob.audioFormat));
- decoder_initialized(decoder);
+ decoder_initialized(decoder, &audio_format);
}
comments = ov_comment(&vf, -1)->user_comments;
putOggCommentsIntoOutputBuffer(inStream->metaName,
@@ -312,7 +311,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
decoder_data(decoder, inStream,
inStream->seekable,
chunk, chunkpos,
- ov_pcm_tell(&vf) / dc.audioFormat.sampleRate,
+ ov_pcm_tell(&vf) / audio_format.sampleRate,
bitRate, replayGainInfo);
chunkpos = 0;
if (dc.command == DECODE_COMMAND_STOP)
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index 9d125f053..167278f01 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -128,6 +128,7 @@ static void wavpack_decode(struct decoder * decoder,
WavpackContext *wpc, int canseek,
ReplayGainInfo *replayGainInfo)
{
+ AudioFormat audio_format;
void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
char chunk[CHUNK_SIZE];
float file_time;
@@ -136,12 +137,12 @@ static void wavpack_decode(struct decoder * decoder,
int position, outsamplesize;
int Bps;
- dc.audioFormat.sampleRate = WavpackGetSampleRate(wpc);
- dc.audioFormat.channels = WavpackGetReducedChannels(wpc);
- dc.audioFormat.bits = WavpackGetBitsPerSample(wpc);
+ audio_format.sampleRate = WavpackGetSampleRate(wpc);
+ audio_format.channels = WavpackGetReducedChannels(wpc);
+ audio_format.bits = WavpackGetBitsPerSample(wpc);
- if (dc.audioFormat.bits > 16)
- dc.audioFormat.bits = 16;
+ if (audio_format.bits > 16)
+ audio_format.bits = 16;
if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
format_samples = format_samples_float;
@@ -159,16 +160,14 @@ static void wavpack_decode(struct decoder * decoder,
outsamplesize = Bps;
if (outsamplesize > 2)
outsamplesize = 2;
- outsamplesize *= dc.audioFormat.channels;
+ outsamplesize *= audio_format.channels;
- samplesreq = sizeof(chunk) / (4 * dc.audioFormat.channels);
+ samplesreq = sizeof(chunk) / (4 * audio_format.channels);
- getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
-
- dc.totalTime = (float)allsamples / dc.audioFormat.sampleRate;
+ dc.totalTime = (float)allsamples / audio_format.sampleRate;
dc.seekable = canseek;
- decoder_initialized(decoder);
+ decoder_initialized(decoder, &audio_format);
position = 0;
@@ -180,7 +179,7 @@ static void wavpack_decode(struct decoder * decoder,
decoder_clear(decoder);
where = dc.seekWhere *
- dc.audioFormat.sampleRate;
+ audio_format.sampleRate;
if (WavpackSeekSample(wpc, where))
position = where;
else
@@ -202,10 +201,10 @@ static void wavpack_decode(struct decoder * decoder,
1000 + 0.5);
position += samplesgot;
file_time = (float)position /
- dc.audioFormat.sampleRate;
+ audio_format.sampleRate;
format_samples(Bps, chunk,
- samplesgot * dc.audioFormat.channels);
+ samplesgot * audio_format.channels);
decoder_data(decoder, NULL, 0, chunk,
samplesgot * outsamplesize,
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-11 14:33:23 +0000