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some more work on organizing code for resampling/audioFormat conversion

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git-svn-id: https://svn.musicpd.org/mpd/trunk@968 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes
2004-05-10 12:35:18 +00:00
parent cd3180c701
commit 2ec1c5ff3c
17 changed files with 100 additions and 87 deletions
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12
src/aac_decode.c
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@@ -251,7 +251,7 @@ int getAacTotalTime(char * file) {
} }
int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) { int aac_decode(OutputBuffer * cb, DecoderControl * dc) {
float time; float time;
float totalTime; float totalTime;
faacDecHandle decoder; faacDecHandle decoder;
@@ -306,9 +306,9 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
return -1; return -1;
} }
af->bits = 16; dc->audioFormat.bits = 16;
cb->totalTime = totalTime; dc->totalTime = totalTime;
time = 0.0; time = 0.0;
@@ -342,8 +342,10 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
#endif #endif
if(dc->start) { if(dc->start) {
af->channels = frameInfo.channels; dc->audioFormat.channels = frameInfo.channels;
af->sampleRate = sampleRate; dc->audioFormat.sampleRate = sampleRate;
getOutputAudioFormat(&(dc->audioFormat),
&(cb->audioFormat));
dc->state = DECODE_STATE_DECODE; dc->state = DECODE_STATE_DECODE;
dc->start = 0; dc->start = 0;
} }

2
src/aac_decode.h
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@@ -27,7 +27,7 @@
int getAacTotalTime(char * file); int getAacTotalTime(char * file);
int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc); int aac_decode(OutputBuffer * cb, DecoderControl * dc);
#endif /* HAVE_FAAD */ #endif /* HAVE_FAAD */

23
src/audiofile_decode.c
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@@ -51,8 +51,7 @@ int getAudiofileTotalTime(char * file)
return time; return time;
} }
int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) int audiofile_decode(OutputBuffer * cb, DecoderControl * dc) {
{
int fs, frame_count; int fs, frame_count;
AFfilehandle af_fp; AFfilehandle af_fp;
int bits; int bits;
@@ -71,19 +70,20 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
} }
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
af->bits = bits; dc->audioFormat.bits = bits;
af->sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK); dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
af->channels = afGetChannels(af_fp,AF_DEFAULT_TRACK); dc->audioFormat.channels = afGetChannels(af_fp,AF_DEFAULT_TRACK);
getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat));
frame_count = afGetFrameCount(af_fp,AF_DEFAULT_TRACK); frame_count = afGetFrameCount(af_fp,AF_DEFAULT_TRACK);
cb->totalTime = ((float)frame_count/(float)af->sampleRate); dc->totalTime = ((float)frame_count/(float)dc->audioFormat.sampleRate);
bitRate = st.st_size*8.0/cb->totalTime/1000.0+0.5; bitRate = st.st_size*8.0/dc->totalTime/1000.0+0.5;
if (af->bits != 8 && af->bits != 16) { if (dc->audioFormat.bits != 8 && dc->audioFormat.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",
dc->file,af->bits); dc->file,dc->audioFormat.bits);
afCloseFile(af_fp); afCloseFile(af_fp);
return -1; return -1;
} }
@@ -100,7 +100,8 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
if(dc->seek) { if(dc->seek) {
cb->end = cb->begin; cb->end = cb->begin;
cb->wrap = 0; cb->wrap = 0;
current = dc->seekWhere * af->sampleRate; current = dc->seekWhere *
dc->audioFormat.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK,current); afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);
dc->seek = 0; dc->seek = 0;
@@ -112,7 +113,7 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
current += ret; current += ret;
sendDataToOutputBuffer(cb,dc,chunk,ret*fs, sendDataToOutputBuffer(cb,dc,chunk,ret*fs,
(float)current / (float)current /
(float)af->sampleRate, (float)dc->audioFormat.sampleRate,
bitRate); bitRate);
if(dc->stop) break; if(dc->stop) break;
else if(dc->seek) continue; else if(dc->seek) continue;

2
src/audiofile_decode.h
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@@ -27,7 +27,7 @@
#include "playerData.h" #include "playerData.h"
int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc); int audiofile_decode(OutputBuffer * cb, DecoderControl * dc);
int getAudiofileTotalTime(char * file); int getAudiofileTotalTime(char * file);

57
src/decode.c
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@@ -111,9 +111,7 @@ int calculateCrossFadeChunks(PlayerControl * pc, AudioFormat * af) {
return (int)chunks; return (int)chunks;
} }
int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc, int waitOnDecode(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb) {
OutputBuffer * cb)
{
while(decode_pid && *decode_pid>0 && dc->start) my_usleep(1000); while(decode_pid && *decode_pid>0 && dc->start) my_usleep(1000);
if(dc->start || dc->error!=DECODE_ERROR_NOERROR) { if(dc->start || dc->error!=DECODE_ERROR_NOERROR) {
@@ -124,7 +122,7 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
return -1; return -1;
} }
if(openAudioDevice(af)<0) { if(openAudioDevice(&(cb->audioFormat))<0) {
strncpy(pc->erroredFile,pc->file,MAXPATHLEN); strncpy(pc->erroredFile,pc->file,MAXPATHLEN);
pc->erroredFile[MAXPATHLEN] = '\0'; pc->erroredFile[MAXPATHLEN] = '\0';
pc->error = PLAYER_ERROR_AUDIO; pc->error = PLAYER_ERROR_AUDIO;
@@ -132,19 +130,17 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
return -1; return -1;
} }
pc->totalTime = cb->totalTime; pc->totalTime = dc->totalTime;
pc->elapsedTime = 0; pc->elapsedTime = 0;
pc->bitRate = 0; pc->bitRate = 0;
pc->sampleRate = af->sampleRate; pc->sampleRate = dc->audioFormat.sampleRate;
pc->bits = af->bits; pc->bits = dc->audioFormat.bits;
pc->channels = af->channels; pc->channels = dc->audioFormat.channels;
return 0; return 0;
} }
void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc, void decodeSeek(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb) {
OutputBuffer * cb)
{
if(decode_pid && *decode_pid>0) { if(decode_pid && *decode_pid>0) {
cb->next = -1; cb->next = -1;
if(dc->state!=DECODE_STATE_DECODE || dc->error || if(dc->state!=DECODE_STATE_DECODE || dc->error ||
@@ -156,7 +152,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
cb->wrap = 0; cb->wrap = 0;
dc->error = 0; dc->error = 0;
dc->start = 1; dc->start = 1;
waitOnDecode(pc,af,dc,cb); waitOnDecode(pc,dc,cb);
} }
if(*decode_pid>0 && dc->state==DECODE_STATE_DECODE) { if(*decode_pid>0 && dc->state==DECODE_STATE_DECODE) {
dc->seekWhere = pc->seekWhere > pc->totalTime-0.1 ? dc->seekWhere = pc->seekWhere > pc->totalTime-0.1 ?
@@ -205,7 +201,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
} \ } \
if(pc->seek) { \ if(pc->seek) { \
pc->totalPlayTime+= pc->elapsedTime-pc->beginTime; \ pc->totalPlayTime+= pc->elapsedTime-pc->beginTime; \
decodeSeek(pc,af,dc,cb); \ decodeSeek(pc,dc,cb); \
pc->beginTime = pc->elapsedTime; \ pc->beginTime = pc->elapsedTime; \
doCrossFade = 0; \ doCrossFade = 0; \
nextChunk = -1; \ nextChunk = -1; \
@@ -217,8 +213,8 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
return; \ return; \
} }
int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af, int decoderInit(PlayerControl * pc, OutputBuffer * cb, DecoderControl * dc) {
DecoderControl * dc) {
int pid; int pid;
int ret; int ret;
decode_pid = &(pc->decode_pid); decode_pid = &(pc->decode_pid);
@@ -237,30 +233,30 @@ int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
switch(pc->decodeType) { switch(pc->decodeType) {
#ifdef HAVE_MAD #ifdef HAVE_MAD
case DECODE_TYPE_MP3: case DECODE_TYPE_MP3:
ret = mp3_decode(cb,af,dc); ret = mp3_decode(cb,dc);
break; break;
#endif #endif
#ifdef HAVE_FAAD #ifdef HAVE_FAAD
case DECODE_TYPE_AAC: case DECODE_TYPE_AAC:
ret = aac_decode(cb,af,dc); ret = aac_decode(cb,dc);
break; break;
case DECODE_TYPE_MP4: case DECODE_TYPE_MP4:
ret = mp4_decode(cb,af,dc); ret = mp4_decode(cb,dc);
break; break;
#endif #endif
#ifdef HAVE_OGG #ifdef HAVE_OGG
case DECODE_TYPE_OGG: case DECODE_TYPE_OGG:
ret = ogg_decode(cb,af,dc); ret = ogg_decode(cb,dc);
break; break;
#endif #endif
#ifdef HAVE_FLAC #ifdef HAVE_FLAC
case DECODE_TYPE_FLAC: case DECODE_TYPE_FLAC:
ret = flac_decode(cb,af,dc); ret = flac_decode(cb,dc);
break; break;
#endif #endif
#ifdef HAVE_AUDIOFILE #ifdef HAVE_AUDIOFILE
case DECODE_TYPE_AUDIOFILE: case DECODE_TYPE_AUDIOFILE:
ret = audiofile_decode(cb,af,dc); ret = audiofile_decode(cb,dc);
break; break;
#endif #endif
default: default:
@@ -313,7 +309,6 @@ int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
void decode() { void decode() {
OutputBuffer * cb; OutputBuffer * cb;
PlayerControl * pc; PlayerControl * pc;
AudioFormat * af;
DecoderControl * dc; DecoderControl * dc;
cb = &(getPlayerData()->buffer); cb = &(getPlayerData()->buffer);
@@ -323,13 +318,12 @@ void decode() {
cb->wrap = 0; cb->wrap = 0;
pc = &(getPlayerData()->playerControl); pc = &(getPlayerData()->playerControl);
dc = &(getPlayerData()->decoderControl); dc = &(getPlayerData()->decoderControl);
af = &(getPlayerData()->audioFormat);
dc->error = 0; dc->error = 0;
dc->start = 1; dc->start = 1;
cb->next = -1; cb->next = -1;
if(decode_pid==NULL || *decode_pid<=0) { if(decode_pid==NULL || *decode_pid<=0) {
if(decoderInit(pc,cb,af,dc)<0) return; if(decoderInit(pc,cb,dc)<0) return;
} }
{ {
@@ -343,7 +337,7 @@ void decode() {
int nextChunk = -1; int nextChunk = -1;
int test; int test;
if(waitOnDecode(pc,af,dc,cb)<0) return; if(waitOnDecode(pc,dc,cb)<0) return;
pc->state = PLAYER_STATE_PLAY; pc->state = PLAYER_STATE_PLAY;
pc->play = 0; pc->play = 0;
@@ -371,12 +365,13 @@ void decode() {
} }
if(cb->next>=0 && doCrossFade==0 && !dc->start) { if(cb->next>=0 && doCrossFade==0 && !dc->start) {
nextChunk = -1; nextChunk = -1;
if(isCurrentAudioFormat(af)) { if(isCurrentAudioFormat(&(cb->audioFormat))) {
doCrossFade = 1; doCrossFade = 1;
crossFadeChunks = crossFadeChunks =
calculateCrossFadeChunks(pc,af); calculateCrossFadeChunks(pc,
&(cb->audioFormat));
if(!crossFadeChunks || if(!crossFadeChunks ||
pc->crossFade>=cb->totalTime) pc->crossFade>=dc->totalTime)
{ {
doCrossFade = -1; doCrossFade = -1;
} }
@@ -415,7 +410,7 @@ void decode() {
cb->begin], cb->begin],
cb->chunkSize[ cb->chunkSize[
nextChunk], nextChunk],
af, &(cb->audioFormat),
((float)fadePosition)/ ((float)fadePosition)/
crossFadeChunks); crossFadeChunks);
if(cb->chunkSize[nextChunk]> if(cb->chunkSize[nextChunk]>
@@ -440,7 +435,7 @@ void decode() {
pcm_volumeChange(cb->chunks+cb->begin* pcm_volumeChange(cb->chunks+cb->begin*
CHUNK_SIZE, CHUNK_SIZE,
cb->chunkSize[cb->begin], cb->chunkSize[cb->begin],
af, &(cb->audioFormat),
pc->softwareVolume); pc->softwareVolume);
if(playAudio(cb->chunks+cb->begin*CHUNK_SIZE, if(playAudio(cb->chunks+cb->begin*CHUNK_SIZE,
cb->chunkSize[cb->begin])<0) cb->chunkSize[cb->begin])<0)
@@ -485,7 +480,7 @@ void decode() {
} }
else { else {
cb->next = -1; cb->next = -1;
if(waitOnDecode(pc,af,dc,cb)<0) return; if(waitOnDecode(pc,dc,cb)<0) return;
nextChunk = -1; nextChunk = -1;
doCrossFade = 0; doCrossFade = 0;
crossFadeChunks = 0; crossFadeChunks = 0;

5
src/decode.h
View File

@@ -22,6 +22,7 @@
#include "../config.h" #include "../config.h"
#include "mpd_types.h" #include "mpd_types.h"
#include "audio.h"
#include <stdio.h> #include <stdio.h>
#include <sys/param.h> #include <sys/param.h>
@@ -48,8 +49,10 @@ typedef struct _DecoderControl {
volatile mpd_sint8 seek; volatile mpd_sint8 seek;
volatile mpd_sint8 seekError; volatile mpd_sint8 seekError;
volatile mpd_sint8 cycleLogFiles; volatile mpd_sint8 cycleLogFiles;
double seekWhere; volatile double seekWhere;
char file[MAXPATHLEN+1]; char file[MAXPATHLEN+1];
AudioFormat audioFormat;
volatile float totalTime;
} DecoderControl; } DecoderControl;
void decodeSigHandler(int sig); void decodeSigHandler(int sig);

29
src/flac_decode.c
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@@ -26,6 +26,7 @@
#include "inputStream.h" #include "inputStream.h"
#include "outputBuffer.h" #include "outputBuffer.h"
#include "replayGain.h" #include "replayGain.h"
#include "audio.h"
#include <stdio.h> #include <stdio.h>
#include <string.h> #include <string.h>
@@ -40,7 +41,6 @@ typedef struct {
int bitRate; int bitRate;
FLAC__uint64 position; FLAC__uint64 position;
OutputBuffer * cb; OutputBuffer * cb;
AudioFormat * af;
DecoderControl * dc; DecoderControl * dc;
InputStream inStream; InputStream inStream;
float replayGainScale; float replayGainScale;
@@ -67,7 +67,7 @@ FLAC__SeekableStreamDecoderLengthStatus flacLength(
const FLAC__SeekableStreamDecoder *, FLAC__uint64 *, void *); const FLAC__SeekableStreamDecoder *, FLAC__uint64 *, void *);
FLAC__bool flacEOF(const FLAC__SeekableStreamDecoder *, void *); FLAC__bool flacEOF(const FLAC__SeekableStreamDecoder *, void *);
int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) { int flac_decode(OutputBuffer * cb, DecoderControl *dc) {
FLAC__SeekableStreamDecoder * flacDec; FLAC__SeekableStreamDecoder * flacDec;
FlacData data; FlacData data;
int status = 1; int status = 1;
@@ -77,7 +77,6 @@ int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
data.position = 0; data.position = 0;
data.bitRate = 0; data.bitRate = 0;
data.cb = cb; data.cb = cb;
data.af = af;
data.dc = dc; data.dc = dc;
data.replayGainScale = 1.0; data.replayGainScale = 1.0;
@@ -146,14 +145,14 @@ int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
} }
if(dc->seek) { if(dc->seek) {
FLAC__uint64 sampleToSeek = dc->seekWhere* FLAC__uint64 sampleToSeek = dc->seekWhere*
af->sampleRate+0.5; dc->audioFormat.sampleRate+0.5;
cb->end = cb->begin; cb->end = cb->begin;
cb->wrap = 0; cb->wrap = 0;
if(FLAC__seekable_stream_decoder_seek_absolute(flacDec, if(FLAC__seekable_stream_decoder_seek_absolute(flacDec,
sampleToSeek)) sampleToSeek))
{ {
data.time = ((float)sampleToSeek)/ data.time = ((float)sampleToSeek)/
af->sampleRate; dc->audioFormat.sampleRate;
data.position = 0; data.position = 0;
} }
dc->seek = 0; dc->seek = 0;
@@ -354,13 +353,17 @@ void flacMetadata(const FLAC__SeekableStreamDecoder *dec,
switch(block->type) { switch(block->type) {
case FLAC__METADATA_TYPE_STREAMINFO: case FLAC__METADATA_TYPE_STREAMINFO:
data->af->bits = block->data.stream_info.bits_per_sample; data->dc->audioFormat.bits =
data->af->bits = 16; block->data.stream_info.bits_per_sample;
data->af->sampleRate = block->data.stream_info.sample_rate; data->dc->audioFormat.sampleRate =
data->af->channels = block->data.stream_info.channels; block->data.stream_info.sample_rate;
data->cb->totalTime = data->dc->audioFormat.channels =
block->data.stream_info.channels;
data->dc->totalTime =
((float)block->data.stream_info.total_samples)/ ((float)block->data.stream_info.total_samples)/
data->af->sampleRate; data->dc->audioFormat.sampleRate;
getOutputAudioFormat(&(data->dc->audioFormat),
&(data->cb->audioFormat));
break; break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT: case FLAC__METADATA_TYPE_VORBIS_COMMENT:
flacParseReplayGain(block,data); flacParseReplayGain(block,data);
@@ -370,7 +373,7 @@ void flacMetadata(const FLAC__SeekableStreamDecoder *dec,
} }
int flacSendChunk(FlacData * data) { int flacSendChunk(FlacData * data) {
doReplayGain(data->chunk,data->chunk_length,data->af, doReplayGain(data->chunk,data->chunk_length,&(data->dc->audioFormat),
data->replayGainScale); data->replayGainScale);
switch(sendDataToOutputBuffer(data->cb,data->dc,data->chunk, switch(sendDataToOutputBuffer(data->cb,data->dc,data->chunk,
@@ -413,7 +416,7 @@ FLAC__StreamDecoderWriteStatus flacWrite(const FLAC__SeekableStreamDecoder *dec,
c_chan++, d_samp++) { c_chan++, d_samp++) {
u16 = buf[c_chan][c_samp]; u16 = buf[c_chan][c_samp];
uc = (unsigned char *)&u16; uc = (unsigned char *)&u16;
for(i=0;i<(data->af->bits/8);i++) { for(i=0;i<(data->dc->audioFormat.bits/8);i++) {
if(data->chunk_length>=CHUNK_SIZE) { if(data->chunk_length>=CHUNK_SIZE) {
if(flacSendChunk(data)<0) { if(flacSendChunk(data)<0) {
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;

2
src/flac_decode.h
View File

@@ -25,7 +25,7 @@
#include <stdio.h> #include <stdio.h>
int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc); int flac_decode(OutputBuffer * cb, DecoderControl * dc);
#endif #endif
/* vim:set shiftwidth=8 tabstop=8 expandtab: */ /* vim:set shiftwidth=8 tabstop=8 expandtab: */

8
src/mp3_decode.c
View File

@@ -523,7 +523,7 @@ void initAudioFormatFromMp3DecodeData(mp3DecodeData * data, AudioFormat * af) {
af->channels = MAD_NCHANNELS(&(data->frame).header); af->channels = MAD_NCHANNELS(&(data->frame).header);
} }
int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) { int mp3_decode(OutputBuffer * cb, DecoderControl * dc) {
mp3DecodeData data; mp3DecodeData data;
if(openMp3(dc->file,&data) < 0) { if(openMp3(dc->file,&data) < 0) {
@@ -531,8 +531,10 @@ int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
return -1; return -1;
} }
initAudioFormatFromMp3DecodeData(&data,af); initAudioFormatFromMp3DecodeData(&data, &(dc->audioFormat));
cb->totalTime = data.totalTime; getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
dc->totalTime = data.totalTime;
dc->start = 0; dc->start = 0;
dc->state = DECODE_STATE_DECODE; dc->state = DECODE_STATE_DECODE;

2
src/mp3_decode.h
View File

@@ -28,7 +28,7 @@
/* this is primarily used in tag.c */ /* this is primarily used in tag.c */
int getMp3TotalTime(char * file); int getMp3TotalTime(char * file);
int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc); int mp3_decode(OutputBuffer * cb, DecoderControl * dc);
#endif #endif

16
src/mp4_decode.c
View File

@@ -84,7 +84,7 @@ uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position) {
} }
int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) { int mp4_decode(OutputBuffer * cb, DecoderControl * dc) {
mp4ff_t * mp4fh; mp4ff_t * mp4fh;
mp4ff_callback_t * mp4cb; mp4ff_callback_t * mp4cb;
int32_t track; int32_t track;
@@ -152,7 +152,7 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
#endif #endif
faacDecSetConfiguration(decoder,config); faacDecSetConfiguration(decoder,config);
af->bits = 16; dc->audioFormat.bits = 16;
mp4Buffer = NULL; mp4Buffer = NULL;
mp4BufferSize = 0; mp4BufferSize = 0;
@@ -169,8 +169,8 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
return -1; return -1;
} }
af->sampleRate = sampleRate; dc->audioFormat.sampleRate = sampleRate;
af->channels = channels; dc->audioFormat.channels = channels;
time = mp4ff_get_track_duration_use_offsets(mp4fh,track); time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
scale = mp4ff_time_scale(mp4fh,track); scale = mp4ff_time_scale(mp4fh,track);
@@ -184,7 +184,7 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
free(mp4cb); free(mp4cb);
return -1; return -1;
} }
cb->totalTime = ((float)time)/scale; dc->totalTime = ((float)time)/scale;
numSamples = mp4ff_num_samples(mp4fh,track); numSamples = mp4ff_num_samples(mp4fh,track);
@@ -255,8 +255,10 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate; scale = frameInfo.samplerate;
#endif #endif
af->sampleRate = scale; dc->audioFormat.sampleRate = scale;
af->channels = frameInfo.channels; dc->audioFormat.channels = frameInfo.channels;
getOutputAudioFormat(&(dc->audioFormat),
&(cb->audioFormat));
dc->state = DECODE_STATE_DECODE; dc->state = DECODE_STATE_DECODE;
dc->start = 0; dc->start = 0;
} }

2
src/mp4_decode.h
View File

@@ -29,7 +29,7 @@
int mp4_getAACTrack(mp4ff_t *infile); int mp4_getAACTrack(mp4ff_t *infile);
int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc); int mp4_decode(OutputBuffer * cb, DecoderControl * dc);
uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer, uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer,
uint32_t length); uint32_t length);

14
src/ogg_decode.c
View File

@@ -142,7 +142,7 @@ float ogg_getReplayGainScale(char ** comments) {
return 1.0; return 1.0;
} }
int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) int ogg_decode(OutputBuffer * cb, DecoderControl * dc)
{ {
OggVorbis_File vf; OggVorbis_File vf;
ov_callbacks callbacks; ov_callbacks callbacks;
@@ -167,12 +167,13 @@ int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
{ {
vorbis_info *vi=ov_info(&vf,-1); vorbis_info *vi=ov_info(&vf,-1);
af->bits = 16; dc->audioFormat.bits = 16;
af->channels = vi->channels; dc->audioFormat.channels = vi->channels;
af->sampleRate = vi->rate; dc->audioFormat.sampleRate = vi->rate;
getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat));
} }
cb->totalTime = ov_time_total(&vf,-1); dc->totalTime = ov_time_total(&vf,-1);
dc->state = DECODE_STATE_DECODE; dc->state = DECODE_STATE_DECODE;
dc->start = 0; dc->start = 0;
@@ -203,7 +204,8 @@ int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
if((test = ov_bitrate_instant(&vf))>0) { if((test = ov_bitrate_instant(&vf))>0) {
bitRate = test/1000; bitRate = test/1000;
} }
doReplayGain(chunk,ret,af,replayGainScale); doReplayGain(chunk,ret,&(dc->audioFormat),
replayGainScale);
sendDataToOutputBuffer(cb,dc,chunk,ret, sendDataToOutputBuffer(cb,dc,chunk,ret,
ov_time_tell(&vf),bitRate); ov_time_tell(&vf),bitRate);
if(dc->stop) break; if(dc->stop) break;

2
src/ogg_decode.h
View File

@@ -25,7 +25,7 @@
#include <stdio.h> #include <stdio.h>
int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc); int ogg_decode(OutputBuffer * cb, DecoderControl * dc);
int getOggTotalTime(char * file); int getOggTotalTime(char * file);

3
src/outputBuffer.h
View File

@@ -21,6 +21,7 @@
#include "mpd_types.h" #include "mpd_types.h"
#include "decode.h" #include "decode.h"
#include "audio.h"
#define OUTPUT_BUFFER_DC_STOP -1 #define OUTPUT_BUFFER_DC_STOP -1
#define OUTPUT_BUFFER_DC_SEEK -2 #define OUTPUT_BUFFER_DC_SEEK -2
@@ -34,7 +35,7 @@ typedef struct _OutputBuffer {
mpd_sint16 volatile end; mpd_sint16 volatile end;
mpd_sint16 volatile next; mpd_sint16 volatile next;
mpd_sint8 volatile wrap; mpd_sint8 volatile wrap;
float totalTime; AudioFormat audioFormat;
} OutputBuffer; } OutputBuffer;
void flushOutputBuffer(OutputBuffer * cb); void flushOutputBuffer(OutputBuffer * cb);

3
src/pcm_utils.h
View File

@@ -33,5 +33,8 @@ void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1, void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
size_t bufferSize2, AudioFormat * format, float portion1); size_t bufferSize2, AudioFormat * format, float portion1);
void pmc_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
inSize, size_t * inLeft, AudioFormat * outFormat,
char * outBuffer, size_t outSize, size_t * outLeft);
#endif #endif
/* vim:set shiftwidth=4 tabstop=8 expandtab: */ /* vim:set shiftwidth=4 tabstop=8 expandtab: */

1
src/playerData.h
View File

@@ -35,7 +35,6 @@ extern int buffered_chunks;
typedef struct _PlayerData { typedef struct _PlayerData {
OutputBuffer buffer; OutputBuffer buffer;
AudioFormat audioFormat;
PlayerControl playerControl; PlayerControl playerControl;
DecoderControl decoderControl; DecoderControl decoderControl;
} PlayerData; } PlayerData;