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added audio_format parameter to decoder_initialized()

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dc->audioFormat is set once by the decoder plugins before invoking
decoder_initialized(); hide dc->audioFormat and let the decoder pass
an AudioFormat pointer to decoder_initialized().
This commit is contained in:
Max Kellermann 2008-08-26 08:27:05 +02:00
parent 0d45870cea
commit 4590a98f0e
14 changed files with 88 additions and 83 deletions
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9
src/decoder_api.c
View File

@ -24,10 +24,17 @@
#include "playerData.h" #include "playerData.h"
#include "gcc.h" #include "gcc.h"
void decoder_initialized(mpd_unused struct decoder * decoder) void decoder_initialized(mpd_unused struct decoder * decoder,
const AudioFormat * audio_format)
{ {
assert(dc.state == DECODE_STATE_START); assert(dc.state == DECODE_STATE_START);
if (audio_format != NULL) {
dc.audioFormat = *audio_format;
getOutputAudioFormat(audio_format,
&(ob.audioFormat));
}
dc.state = DECODE_STATE_DECODE; dc.state = DECODE_STATE_DECODE;
notify_signal(&pc.notify); notify_signal(&pc.notify);
} }

3
src/decoder_api.h
View File

@ -40,7 +40,8 @@ struct decoder;
* Notify the player thread that it has finished initialization and * Notify the player thread that it has finished initialization and
* that it has read the song's meta data. * that it has read the song's meta data.
*/ */
void decoder_initialized(struct decoder * decoder); void decoder_initialized(struct decoder * decoder,
const AudioFormat * audio_format);
/** /**
* This function is called by the decoder plugin when it has * This function is called by the decoder plugin when it has

7
src/inputPlugins/_flac_common.c
View File

@ -162,11 +162,10 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) { switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO: case FLAC__METADATA_TYPE_STREAMINFO:
dc.audioFormat.bits = (mpd_sint8)si->bits_per_sample; data->audio_format.bits = (mpd_sint8)si->bits_per_sample;
dc.audioFormat.sampleRate = si->sample_rate; data->audio_format.sampleRate = si->sample_rate;
dc.audioFormat.channels = (mpd_sint8)si->channels; data->audio_format.channels = (mpd_sint8)si->channels;
dc.totalTime = ((float)si->total_samples) / (si->sample_rate); dc.totalTime = ((float)si->total_samples) / (si->sample_rate);
getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
break; break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT: case FLAC__METADATA_TYPE_VORBIS_COMMENT:
flacParseReplayGain(block, data); flacParseReplayGain(block, data);

1
src/inputPlugins/_flac_common.h
View File

@ -143,6 +143,7 @@ typedef struct {
size_t chunk_length; size_t chunk_length;
float time; float time;
unsigned int bitRate; unsigned int bitRate;
AudioFormat audio_format;
FLAC__uint64 position; FLAC__uint64 position;
struct decoder *decoder; struct decoder *decoder;
InputStream *inStream; InputStream *inStream;

11
src/inputPlugins/aac_plugin.c
View File

@ -286,6 +286,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
faacDecFrameInfo frameInfo; faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config; faacDecConfigurationPtr config;
long bread; long bread;
AudioFormat audio_format;
uint32_t sampleRate; uint32_t sampleRate;
unsigned char channels; unsigned char channels;
int eof = 0; int eof = 0;
@ -335,7 +336,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
return -1; return -1;
} }
dc.audioFormat.bits = 16; audio_format.bits = 16;
dc.totalTime = totalTime; dc.totalTime = totalTime;
@ -369,11 +370,9 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
#endif #endif
if (dc.state != DECODE_STATE_DECODE) { if (dc.state != DECODE_STATE_DECODE) {
dc.audioFormat.channels = frameInfo.channels; audio_format.channels = frameInfo.channels;
dc.audioFormat.sampleRate = sampleRate; audio_format.sampleRate = sampleRate;
getOutputAudioFormat(&(dc.audioFormat), decoder_initialized(mpd_decoder, &audio_format);
&(ob.audioFormat));
decoder_initialized(mpd_decoder);
} }
advanceAacBuffer(&b, frameInfo.bytesconsumed); advanceAacBuffer(&b, frameInfo.bytesconsumed);

21
src/inputPlugins/audiofile_plugin.c
View File

@ -45,6 +45,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
int fs, frame_count; int fs, frame_count;
AFfilehandle af_fp; AFfilehandle af_fp;
int bits; int bits;
AudioFormat audio_format;
mpd_uint16 bitRate; mpd_uint16 bitRate;
struct stat st; struct stat st;
@ -62,30 +63,30 @@ static int audiofile_decode(struct decoder * decoder, char *path)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16); AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
dc.audioFormat.bits = (mpd_uint8)bits; audio_format.bits = (mpd_uint8)bits;
dc.audioFormat.sampleRate = audio_format.sampleRate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
dc.audioFormat.channels = audio_format.channels =
(mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
dc.totalTime = 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); 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", 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); afCloseFile(af_fp);
return -1; return -1;
} }
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder); decoder_initialized(decoder, &audio_format);
{ {
int ret, eof = 0, current = 0; int ret, eof = 0, current = 0;
char chunk[CHUNK_SIZE]; char chunk[CHUNK_SIZE];
@ -94,7 +95,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
if (dc.command == DECODE_COMMAND_SEEK) { if (dc.command == DECODE_COMMAND_SEEK) {
decoder_clear(decoder); decoder_clear(decoder);
current = dc.seekWhere * current = dc.seekWhere *
dc.audioFormat.sampleRate; audio_format.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
dc_command_finished(); dc_command_finished();
} }
@ -110,7 +111,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
1, 1,
chunk, ret * fs, chunk, ret * fs,
(float)current / (float)current /
(float)dc.audioFormat. (float)audio_format.
sampleRate, bitRate, sampleRate, bitRate,
NULL); NULL);
if (dc.command == DECODE_COMMAND_STOP) if (dc.command == DECODE_COMMAND_STOP)

10
src/inputPlugins/flac_plugin.c
View File

@ -242,7 +242,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
FLAC__uint32 samples = frame->header.blocksize; FLAC__uint32 samples = frame->header.blocksize;
unsigned int c_samp; unsigned int c_samp;
const unsigned int num_channels = frame->header.channels; 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 = const unsigned int bytes_per_channel =
bytes_per_sample * frame->header.channels; bytes_per_sample * frame->header.channels;
const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel; 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; float timeChange;
FLAC__uint64 newPosition = 0; FLAC__uint64 newPosition = 0;
assert(dc.audioFormat.bits > 0); assert(data->audio_format.bits > 0);
timeChange = ((float)samples) / frame->header.sample_rate; timeChange = ((float)samples) / frame->header.sample_rate;
data->time += timeChange; 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) { while (1) {
if (!flac_process_single(flacDec)) if (!flac_process_single(flacDec))
@ -424,11 +424,11 @@ static int flac_decode_internal(struct decoder * decoder,
break; break;
if (dc.command == DECODE_COMMAND_SEEK) { if (dc.command == DECODE_COMMAND_SEEK) {
FLAC__uint64 sampleToSeek = dc.seekWhere * FLAC__uint64 sampleToSeek = dc.seekWhere *
dc.audioFormat.sampleRate + 0.5; data.audio_format.sampleRate + 0.5;
if (flac_seek_absolute(flacDec, sampleToSeek)) { if (flac_seek_absolute(flacDec, sampleToSeek)) {
decoder_clear(decoder); decoder_clear(decoder);
data.time = ((float)sampleToSeek) / data.time = ((float)sampleToSeek) /
dc.audioFormat.sampleRate; data.audio_format.sampleRate;
data.position = 0; data.position = 0;
} else } else
dc.seekError = 1; dc.seekError = 1;

15
src/inputPlugins/mod_plugin.c
View File

@ -162,6 +162,7 @@ static void mod_close(mod_Data * data)
static int mod_decode(struct decoder * decoder, char *path) static int mod_decode(struct decoder * decoder, char *path)
{ {
mod_Data *data; mod_Data *data;
AudioFormat audio_format;
float total_time = 0.0; float total_time = 0.0;
int ret; int ret;
float secPerByte; float secPerByte;
@ -176,16 +177,16 @@ static int mod_decode(struct decoder * decoder, char *path)
} }
dc.totalTime = 0; dc.totalTime = 0;
dc.audioFormat.bits = 16; audio_format.bits = 16;
dc.audioFormat.sampleRate = 44100; audio_format.sampleRate = 44100;
dc.audioFormat.channels = 2; audio_format.channels = 2;
getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
secPerByte = secPerByte =
1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) * 1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
(float)dc.audioFormat.sampleRate); (float)audio_format.sampleRate);
decoder_initialized(decoder, &audio_format);
decoder_initialized(decoder);
while (1) { while (1) {
if (dc.command == DECODE_COMMAND_SEEK) { if (dc.command == DECODE_COMMAND_SEEK) {
dc.seekError = 1; dc.seekError = 1;

6
src/inputPlugins/mp3_plugin.c
View File

@ -1021,6 +1021,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
mp3DecodeData data; mp3DecodeData data;
MpdTag *tag = NULL; MpdTag *tag = NULL;
ReplayGainInfo *replayGainInfo = NULL; ReplayGainInfo *replayGainInfo = NULL;
AudioFormat audio_format;
if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) < if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) <
0) { 0) {
@ -1032,8 +1033,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
return 0; return 0;
} }
initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat)); initAudioFormatFromMp3DecodeData(&data, &audio_format);
getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
dc.totalTime = data.totalTime; dc.totalTime = data.totalTime;
@ -1062,7 +1062,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
freeMpdTag(tag); freeMpdTag(tag);
} }
decoder_initialized(decoder); decoder_initialized(decoder, &audio_format);
while (mp3Read(&data, decoder, &replayGainInfo) != DECODE_BREAK) ; while (mp3Read(&data, decoder, &replayGainInfo) != DECODE_BREAK) ;
/* send last little bit if not DECODE_COMMAND_STOP */ /* send last little bit if not DECODE_COMMAND_STOP */

15
src/inputPlugins/mp4_plugin.c
View File

@ -88,6 +88,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
faacDecHandle decoder; faacDecHandle decoder;
faacDecFrameInfo frameInfo; faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config; faacDecConfigurationPtr config;
AudioFormat audio_format;
unsigned char *mp4Buffer; unsigned char *mp4Buffer;
unsigned int mp4BufferSize; unsigned int mp4BufferSize;
uint32_t sampleRate; uint32_t sampleRate;
@ -139,7 +140,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
#endif #endif
faacDecSetConfiguration(decoder, config); faacDecSetConfiguration(decoder, config);
dc.audioFormat.bits = 16; audio_format.bits = 16;
mp4Buffer = NULL; mp4Buffer = NULL;
mp4BufferSize = 0; mp4BufferSize = 0;
@ -154,8 +155,8 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
return -1; return -1;
} }
dc.audioFormat.sampleRate = sampleRate; audio_format.sampleRate = sampleRate;
dc.audioFormat.channels = channels; audio_format.channels = channels;
file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track); file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(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 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate; scale = frameInfo.samplerate;
#endif #endif
dc.audioFormat.sampleRate = scale; audio_format.sampleRate = scale;
dc.audioFormat.channels = frameInfo.channels; audio_format.channels = frameInfo.channels;
getOutputAudioFormat(&(dc.audioFormat), decoder_initialized(mpd_decoder, &audio_format);
&(ob.audioFormat));
decoder_initialized(mpd_decoder);
} }
if (channels * (unsigned long)(dur + offset) > frameInfo.samples) { if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {

21
src/inputPlugins/mpc_plugin.c
View File

@ -111,6 +111,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
mpc_decoder decoder; mpc_decoder decoder;
mpc_reader reader; mpc_reader reader;
mpc_streaminfo info; mpc_streaminfo info;
AudioFormat audio_format;
MpcCallbackData data; MpcCallbackData data;
@ -161,11 +162,9 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
dc.totalTime = mpc_streaminfo_get_length(&info); dc.totalTime = mpc_streaminfo_get_length(&info);
dc.audioFormat.bits = 16; audio_format.bits = 16;
dc.audioFormat.channels = info.channels; audio_format.channels = info.channels;
dc.audioFormat.sampleRate = info.sample_freq; audio_format.sampleRate = info.sample_freq;
getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
replayGainInfo = newReplayGainInfo(); replayGainInfo = newReplayGainInfo();
replayGainInfo->albumGain = info.gain_album * 0.01; 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->trackGain = info.gain_title * 0.01;
replayGainInfo->trackPeak = info.peak_title / 32767.0; replayGainInfo->trackPeak = info.peak_title / 32767.0;
decoder_initialized(mpd_decoder); decoder_initialized(mpd_decoder, &audio_format);
while (!eof) { while (!eof) {
if (dc.command == DECODE_COMMAND_SEEK) { 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)) { if (mpc_decoder_seek_sample(&decoder, samplePos)) {
decoder_clear(mpd_decoder); decoder_clear(mpd_decoder);
s16 = (mpd_sint16 *) chunk; s16 = (mpd_sint16 *) chunk;
@ -210,10 +209,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
if (chunkpos >= MPC_CHUNK_SIZE) { if (chunkpos >= MPC_CHUNK_SIZE) {
total_time = ((float)samplePos) / total_time = ((float)samplePos) /
dc.audioFormat.sampleRate; audio_format.sampleRate;
bitRate = vbrUpdateBits * bitRate = vbrUpdateBits *
dc.audioFormat.sampleRate / 1152 / 1000; audio_format.sampleRate / 1152 / 1000;
decoder_data(mpd_decoder, inStream, decoder_data(mpd_decoder, inStream,
inStream->seekable, inStream->seekable,
@ -232,10 +231,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
} }
if (dc.command != DECODE_COMMAND_STOP && chunkpos > 0) { if (dc.command != DECODE_COMMAND_STOP && chunkpos > 0) {
total_time = ((float)samplePos) / dc.audioFormat.sampleRate; total_time = ((float)samplePos) / audio_format.sampleRate;
bitRate = bitRate =
vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000; vbrUpdateBits * audio_format.sampleRate / 1152 / 1000;
decoder_data(mpd_decoder, NULL, inStream->seekable, decoder_data(mpd_decoder, NULL, inStream->seekable,
chunk, chunkpos, total_time, bitRate, chunk, chunkpos, total_time, bitRate,

12
src/inputPlugins/oggflac_plugin.c
View File

@ -188,7 +188,7 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(const
c_chan++) { c_chan++) {
u16 = buf[c_chan][c_samp]; u16 = buf[c_chan][c_samp];
uc = (unsigned char *)&u16; 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 (data->chunk_length >= FLAC_CHUNK_SIZE) {
if (flacSendChunk(data) < 0) { if (flacSendChunk(data) < 0) {
return return
@ -345,7 +345,7 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
goto fail; goto fail;
} }
decoder_initialized(mpd_decoder); decoder_initialized(mpd_decoder, &data.audio_format);
while (1) { while (1) {
OggFLAC__seekable_stream_decoder_process_single(decoder); 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) { OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
break; break;
} }
if (dc.command == DECODE_COMMAND_SEEK) { if (dc->command == DECODE_COMMAND_SEEK) {
FLAC__uint64 sampleToSeek = dc.seekWhere * FLAC__uint64 sampleToSeek = dc->seekWhere *
dc.audioFormat.sampleRate + 0.5; data.audio_format.sampleRate + 0.5;
if (OggFLAC__seekable_stream_decoder_seek_absolute if (OggFLAC__seekable_stream_decoder_seek_absolute
(decoder, sampleToSeek)) { (decoder, sampleToSeek)) {
decoder_clear(mpd_decoder); decoder_clear(mpd_decoder);
data.time = ((float)sampleToSeek) / data.time = ((float)sampleToSeek) /
dc.audioFormat.sampleRate; data.audio_format.sampleRate;
data.position = 0; data.position = 0;
} else } else
dc.seekError = 1; dc.seekError = 1;

13
src/inputPlugins/oggvorbis_plugin.c
View File

@ -215,6 +215,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
OggVorbis_File vf; OggVorbis_File vf;
ov_callbacks callbacks; ov_callbacks callbacks;
OggCallbackData data; OggCallbackData data;
AudioFormat audio_format;
int current_section; int current_section;
int prev_section = -1; int prev_section = -1;
long ret; long ret;
@ -264,7 +265,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
dc.totalTime = ov_time_total(&vf, -1); dc.totalTime = ov_time_total(&vf, -1);
if (dc.totalTime < 0) if (dc.totalTime < 0)
dc.totalTime = 0; dc.totalTime = 0;
dc.audioFormat.bits = 16; audio_format.bits = 16;
while (1) { while (1) {
if (dc.command == DECODE_COMMAND_SEEK) { if (dc.command == DECODE_COMMAND_SEEK) {
@ -281,12 +282,10 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
if (current_section != prev_section) { if (current_section != prev_section) {
/*printf("new song!\n"); */ /*printf("new song!\n"); */
vorbis_info *vi = ov_info(&vf, -1); vorbis_info *vi = ov_info(&vf, -1);
dc.audioFormat.channels = vi->channels; audio_format.channels = vi->channels;
dc.audioFormat.sampleRate = vi->rate; audio_format.sampleRate = vi->rate;
if (dc.state == DECODE_STATE_START) { if (dc.state == DECODE_STATE_START) {
getOutputAudioFormat(&(dc.audioFormat), decoder_initialized(decoder, &audio_format);
&(ob.audioFormat));
decoder_initialized(decoder);
} }
comments = ov_comment(&vf, -1)->user_comments; comments = ov_comment(&vf, -1)->user_comments;
putOggCommentsIntoOutputBuffer(inStream->metaName, putOggCommentsIntoOutputBuffer(inStream->metaName,
@ -312,7 +311,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
decoder_data(decoder, inStream, decoder_data(decoder, inStream,
inStream->seekable, inStream->seekable,
chunk, chunkpos, chunk, chunkpos,
ov_pcm_tell(&vf) / dc.audioFormat.sampleRate, ov_pcm_tell(&vf) / audio_format.sampleRate,
bitRate, replayGainInfo); bitRate, replayGainInfo);
chunkpos = 0; chunkpos = 0;
if (dc.command == DECODE_COMMAND_STOP) if (dc.command == DECODE_COMMAND_STOP)

27
src/inputPlugins/wavpack_plugin.c
View File

@ -128,6 +128,7 @@ static void wavpack_decode(struct decoder * decoder,
WavpackContext *wpc, int canseek, WavpackContext *wpc, int canseek,
ReplayGainInfo *replayGainInfo) ReplayGainInfo *replayGainInfo)
{ {
AudioFormat audio_format;
void (*format_samples)(int Bps, void *buffer, uint32_t samcnt); void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
char chunk[CHUNK_SIZE]; char chunk[CHUNK_SIZE];
float file_time; float file_time;
@ -136,12 +137,12 @@ static void wavpack_decode(struct decoder * decoder,
int position, outsamplesize; int position, outsamplesize;
int Bps; int Bps;
dc.audioFormat.sampleRate = WavpackGetSampleRate(wpc); audio_format.sampleRate = WavpackGetSampleRate(wpc);
dc.audioFormat.channels = WavpackGetReducedChannels(wpc); audio_format.channels = WavpackGetReducedChannels(wpc);
dc.audioFormat.bits = WavpackGetBitsPerSample(wpc); audio_format.bits = WavpackGetBitsPerSample(wpc);
if (dc.audioFormat.bits > 16) if (audio_format.bits > 16)
dc.audioFormat.bits = 16; audio_format.bits = 16;
if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT) if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
format_samples = format_samples_float; format_samples = format_samples_float;
@ -159,16 +160,14 @@ static void wavpack_decode(struct decoder * decoder,
outsamplesize = Bps; outsamplesize = Bps;
if (outsamplesize > 2) if (outsamplesize > 2)
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 / audio_format.sampleRate;
dc.totalTime = (float)allsamples / dc.audioFormat.sampleRate;
dc.seekable = canseek; dc.seekable = canseek;
decoder_initialized(decoder); decoder_initialized(decoder, &audio_format);
position = 0; position = 0;
@ -180,7 +179,7 @@ static void wavpack_decode(struct decoder * decoder,
decoder_clear(decoder); decoder_clear(decoder);
where = dc.seekWhere * where = dc.seekWhere *
dc.audioFormat.sampleRate; audio_format.sampleRate;
if (WavpackSeekSample(wpc, where)) if (WavpackSeekSample(wpc, where))
position = where; position = where;
else else
@ -202,10 +201,10 @@ static void wavpack_decode(struct decoder * decoder,
1000 + 0.5); 1000 + 0.5);
position += samplesgot; position += samplesgot;
file_time = (float)position / file_time = (float)position /
dc.audioFormat.sampleRate; audio_format.sampleRate;
format_samples(Bps, chunk, format_samples(Bps, chunk,
samplesgot * dc.audioFormat.channels); samplesgot * audio_format.channels);
decoder_data(decoder, NULL, 0, chunk, decoder_data(decoder, NULL, 0, chunk,
samplesgot * outsamplesize, samplesgot * outsamplesize,