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-rw-r--r--src/aac_decode.c12
-rw-r--r--src/aac_decode.h2
-rw-r--r--src/audiofile_decode.c27
-rw-r--r--src/audiofile_decode.h2
-rw-r--r--src/decode.c57
-rw-r--r--src/decode.h5
-rw-r--r--src/flac_decode.c29
-rw-r--r--src/flac_decode.h2
-rw-r--r--src/mp3_decode.c8
-rw-r--r--src/mp3_decode.h2
-rw-r--r--src/mp4_decode.c16
-rw-r--r--src/mp4_decode.h2
-rw-r--r--src/ogg_decode.c14
-rw-r--r--src/ogg_decode.h2
-rw-r--r--src/outputBuffer.h3
-rw-r--r--src/pcm_utils.h3
-rw-r--r--src/playerData.h1
17 files changed, 100 insertions, 87 deletions
diff --git a/src/aac_decode.c b/src/aac_decode.c
index 24171adb7..7013502d4 100644
--- a/src/aac_decode.c
+++ b/src/aac_decode.c
@@ -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 totalTime;
faacDecHandle decoder;
@@ -306,9 +306,9 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
return -1;
}
- af->bits = 16;
+ dc->audioFormat.bits = 16;
- cb->totalTime = totalTime;
+ dc->totalTime = totalTime;
time = 0.0;
@@ -342,8 +342,10 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
#endif
if(dc->start) {
- af->channels = frameInfo.channels;
- af->sampleRate = sampleRate;
+ dc->audioFormat.channels = frameInfo.channels;
+ dc->audioFormat.sampleRate = sampleRate;
+ getOutputAudioFormat(&(dc->audioFormat),
+ &(cb->audioFormat));
dc->state = DECODE_STATE_DECODE;
dc->start = 0;
}
diff --git a/src/aac_decode.h b/src/aac_decode.h
index 62e77eb1e..3d1636726 100644
--- a/src/aac_decode.h
+++ b/src/aac_decode.h
@@ -27,7 +27,7 @@
int getAacTotalTime(char * file);
-int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int aac_decode(OutputBuffer * cb, DecoderControl * dc);
#endif /* HAVE_FAAD */
diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c
index f3e88395b..b6b459f08 100644
--- a/src/audiofile_decode.c
+++ b/src/audiofile_decode.c
@@ -51,8 +51,7 @@ int getAudiofileTotalTime(char * file)
return time;
}
-int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
-{
+int audiofile_decode(OutputBuffer * cb, DecoderControl * dc) {
int fs, frame_count;
AFfilehandle af_fp;
int bits;
@@ -71,19 +70,20 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
}
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- af->bits = bits;
- af->sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
- af->channels = afGetChannels(af_fp,AF_DEFAULT_TRACK);
+ dc->audioFormat.bits = bits;
+ dc->audioFormat.sampleRate = afGetRate(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);
- 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",
- dc->file,af->bits);
+ dc->file,dc->audioFormat.bits);
afCloseFile(af_fp);
return -1;
}
@@ -100,7 +100,8 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
if(dc->seek) {
cb->end = cb->begin;
cb->wrap = 0;
- current = dc->seekWhere * af->sampleRate;
+ current = dc->seekWhere *
+ dc->audioFormat.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);
dc->seek = 0;
@@ -111,9 +112,9 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
else {
current += ret;
sendDataToOutputBuffer(cb,dc,chunk,ret*fs,
- (float)current /
- (float)af->sampleRate,
- bitRate);
+ (float)current /
+ (float)dc->audioFormat.sampleRate,
+ bitRate);
if(dc->stop) break;
else if(dc->seek) continue;
}
diff --git a/src/audiofile_decode.h b/src/audiofile_decode.h
index 774d58c1b..82dc565ec 100644
--- a/src/audiofile_decode.h
+++ b/src/audiofile_decode.h
@@ -27,7 +27,7 @@
#include "playerData.h"
-int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int audiofile_decode(OutputBuffer * cb, DecoderControl * dc);
int getAudiofileTotalTime(char * file);
diff --git a/src/decode.c b/src/decode.c
index d386a201a..8ab215a4a 100644
--- a/src/decode.c
+++ b/src/decode.c
@@ -111,9 +111,7 @@ int calculateCrossFadeChunks(PlayerControl * pc, AudioFormat * af) {
return (int)chunks;
}
-int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
- OutputBuffer * cb)
-{
+int waitOnDecode(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb) {
while(decode_pid && *decode_pid>0 && dc->start) my_usleep(1000);
if(dc->start || dc->error!=DECODE_ERROR_NOERROR) {
@@ -124,7 +122,7 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
return -1;
}
- if(openAudioDevice(af)<0) {
+ if(openAudioDevice(&(cb->audioFormat))<0) {
strncpy(pc->erroredFile,pc->file,MAXPATHLEN);
pc->erroredFile[MAXPATHLEN] = '\0';
pc->error = PLAYER_ERROR_AUDIO;
@@ -132,19 +130,17 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
return -1;
}
- pc->totalTime = cb->totalTime;
+ pc->totalTime = dc->totalTime;
pc->elapsedTime = 0;
pc->bitRate = 0;
- pc->sampleRate = af->sampleRate;
- pc->bits = af->bits;
- pc->channels = af->channels;
+ pc->sampleRate = dc->audioFormat.sampleRate;
+ pc->bits = dc->audioFormat.bits;
+ pc->channels = dc->audioFormat.channels;
return 0;
}
-void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
- OutputBuffer * cb)
-{
+void decodeSeek(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb) {
if(decode_pid && *decode_pid>0) {
cb->next = -1;
if(dc->state!=DECODE_STATE_DECODE || dc->error ||
@@ -156,7 +152,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
cb->wrap = 0;
dc->error = 0;
dc->start = 1;
- waitOnDecode(pc,af,dc,cb);
+ waitOnDecode(pc,dc,cb);
}
if(*decode_pid>0 && dc->state==DECODE_STATE_DECODE) {
dc->seekWhere = pc->seekWhere > pc->totalTime-0.1 ?
@@ -205,7 +201,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
} \
if(pc->seek) { \
pc->totalPlayTime+= pc->elapsedTime-pc->beginTime; \
- decodeSeek(pc,af,dc,cb); \
+ decodeSeek(pc,dc,cb); \
pc->beginTime = pc->elapsedTime; \
doCrossFade = 0; \
nextChunk = -1; \
@@ -217,8 +213,8 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
return; \
}
-int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
- DecoderControl * dc) {
+int decoderInit(PlayerControl * pc, OutputBuffer * cb, DecoderControl * dc) {
+
int pid;
int ret;
decode_pid = &(pc->decode_pid);
@@ -237,30 +233,30 @@ int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
switch(pc->decodeType) {
#ifdef HAVE_MAD
case DECODE_TYPE_MP3:
- ret = mp3_decode(cb,af,dc);
+ ret = mp3_decode(cb,dc);
break;
#endif
#ifdef HAVE_FAAD
case DECODE_TYPE_AAC:
- ret = aac_decode(cb,af,dc);
+ ret = aac_decode(cb,dc);
break;
case DECODE_TYPE_MP4:
- ret = mp4_decode(cb,af,dc);
+ ret = mp4_decode(cb,dc);
break;
#endif
#ifdef HAVE_OGG
case DECODE_TYPE_OGG:
- ret = ogg_decode(cb,af,dc);
+ ret = ogg_decode(cb,dc);
break;
#endif
#ifdef HAVE_FLAC
case DECODE_TYPE_FLAC:
- ret = flac_decode(cb,af,dc);
+ ret = flac_decode(cb,dc);
break;
#endif
#ifdef HAVE_AUDIOFILE
case DECODE_TYPE_AUDIOFILE:
- ret = audiofile_decode(cb,af,dc);
+ ret = audiofile_decode(cb,dc);
break;
#endif
default:
@@ -313,7 +309,6 @@ int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
void decode() {
OutputBuffer * cb;
PlayerControl * pc;
- AudioFormat * af;
DecoderControl * dc;
cb = &(getPlayerData()->buffer);
@@ -323,13 +318,12 @@ void decode() {
cb->wrap = 0;
pc = &(getPlayerData()->playerControl);
dc = &(getPlayerData()->decoderControl);
- af = &(getPlayerData()->audioFormat);
dc->error = 0;
dc->start = 1;
cb->next = -1;
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 test;
- if(waitOnDecode(pc,af,dc,cb)<0) return;
+ if(waitOnDecode(pc,dc,cb)<0) return;
pc->state = PLAYER_STATE_PLAY;
pc->play = 0;
@@ -371,12 +365,13 @@ void decode() {
}
if(cb->next>=0 && doCrossFade==0 && !dc->start) {
nextChunk = -1;
- if(isCurrentAudioFormat(af)) {
+ if(isCurrentAudioFormat(&(cb->audioFormat))) {
doCrossFade = 1;
crossFadeChunks =
- calculateCrossFadeChunks(pc,af);
+ calculateCrossFadeChunks(pc,
+ &(cb->audioFormat));
if(!crossFadeChunks ||
- pc->crossFade>=cb->totalTime)
+ pc->crossFade>=dc->totalTime)
{
doCrossFade = -1;
}
@@ -415,7 +410,7 @@ void decode() {
cb->begin],
cb->chunkSize[
nextChunk],
- af,
+ &(cb->audioFormat),
((float)fadePosition)/
crossFadeChunks);
if(cb->chunkSize[nextChunk]>
@@ -440,7 +435,7 @@ void decode() {
pcm_volumeChange(cb->chunks+cb->begin*
CHUNK_SIZE,
cb->chunkSize[cb->begin],
- af,
+ &(cb->audioFormat),
pc->softwareVolume);
if(playAudio(cb->chunks+cb->begin*CHUNK_SIZE,
cb->chunkSize[cb->begin])<0)
@@ -485,7 +480,7 @@ void decode() {
}
else {
cb->next = -1;
- if(waitOnDecode(pc,af,dc,cb)<0) return;
+ if(waitOnDecode(pc,dc,cb)<0) return;
nextChunk = -1;
doCrossFade = 0;
crossFadeChunks = 0;
diff --git a/src/decode.h b/src/decode.h
index 8d4168a62..39c6ec34d 100644
--- a/src/decode.h
+++ b/src/decode.h
@@ -22,6 +22,7 @@
#include "../config.h"
#include "mpd_types.h"
+#include "audio.h"
#include <stdio.h>
#include <sys/param.h>
@@ -48,8 +49,10 @@ typedef struct _DecoderControl {
volatile mpd_sint8 seek;
volatile mpd_sint8 seekError;
volatile mpd_sint8 cycleLogFiles;
- double seekWhere;
+ volatile double seekWhere;
char file[MAXPATHLEN+1];
+ AudioFormat audioFormat;
+ volatile float totalTime;
} DecoderControl;
void decodeSigHandler(int sig);
diff --git a/src/flac_decode.c b/src/flac_decode.c
index 7865d700a..93c114068 100644
--- a/src/flac_decode.c
+++ b/src/flac_decode.c
@@ -26,6 +26,7 @@
#include "inputStream.h"
#include "outputBuffer.h"
#include "replayGain.h"
+#include "audio.h"
#include <stdio.h>
#include <string.h>
@@ -40,7 +41,6 @@ typedef struct {
int bitRate;
FLAC__uint64 position;
OutputBuffer * cb;
- AudioFormat * af;
DecoderControl * dc;
InputStream inStream;
float replayGainScale;
@@ -67,7 +67,7 @@ FLAC__SeekableStreamDecoderLengthStatus flacLength(
const FLAC__SeekableStreamDecoder *, FLAC__uint64 *, 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;
FlacData data;
int status = 1;
@@ -77,7 +77,6 @@ int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
data.position = 0;
data.bitRate = 0;
data.cb = cb;
- data.af = af;
data.dc = dc;
data.replayGainScale = 1.0;
@@ -146,14 +145,14 @@ int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
}
if(dc->seek) {
FLAC__uint64 sampleToSeek = dc->seekWhere*
- af->sampleRate+0.5;
+ dc->audioFormat.sampleRate+0.5;
cb->end = cb->begin;
cb->wrap = 0;
if(FLAC__seekable_stream_decoder_seek_absolute(flacDec,
sampleToSeek))
{
data.time = ((float)sampleToSeek)/
- af->sampleRate;
+ dc->audioFormat.sampleRate;
data.position = 0;
}
dc->seek = 0;
@@ -354,13 +353,17 @@ void flacMetadata(const FLAC__SeekableStreamDecoder *dec,
switch(block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- data->af->bits = block->data.stream_info.bits_per_sample;
- data->af->bits = 16;
- data->af->sampleRate = block->data.stream_info.sample_rate;
- data->af->channels = block->data.stream_info.channels;
- data->cb->totalTime =
+ data->dc->audioFormat.bits =
+ block->data.stream_info.bits_per_sample;
+ data->dc->audioFormat.sampleRate =
+ block->data.stream_info.sample_rate;
+ data->dc->audioFormat.channels =
+ block->data.stream_info.channels;
+ data->dc->totalTime =
((float)block->data.stream_info.total_samples)/
- data->af->sampleRate;
+ data->dc->audioFormat.sampleRate;
+ getOutputAudioFormat(&(data->dc->audioFormat),
+ &(data->cb->audioFormat));
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
flacParseReplayGain(block,data);
@@ -370,7 +373,7 @@ void flacMetadata(const FLAC__SeekableStreamDecoder *dec,
}
int flacSendChunk(FlacData * data) {
- doReplayGain(data->chunk,data->chunk_length,data->af,
+ doReplayGain(data->chunk,data->chunk_length,&(data->dc->audioFormat),
data->replayGainScale);
switch(sendDataToOutputBuffer(data->cb,data->dc,data->chunk,
@@ -413,7 +416,7 @@ FLAC__StreamDecoderWriteStatus flacWrite(const FLAC__SeekableStreamDecoder *dec,
c_chan++, d_samp++) {
u16 = buf[c_chan][c_samp];
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(flacSendChunk(data)<0) {
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
diff --git a/src/flac_decode.h b/src/flac_decode.h
index 4858274f8..c7956ca12 100644
--- a/src/flac_decode.h
+++ b/src/flac_decode.h
@@ -25,7 +25,7 @@
#include <stdio.h>
-int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int flac_decode(OutputBuffer * cb, DecoderControl * dc);
#endif
/* vim:set shiftwidth=8 tabstop=8 expandtab: */
diff --git a/src/mp3_decode.c b/src/mp3_decode.c
index d3a6a592e..1b83f6eed 100644
--- a/src/mp3_decode.c
+++ b/src/mp3_decode.c
@@ -523,7 +523,7 @@ void initAudioFormatFromMp3DecodeData(mp3DecodeData * data, AudioFormat * af) {
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;
if(openMp3(dc->file,&data) < 0) {
@@ -531,8 +531,10 @@ int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
return -1;
}
- initAudioFormatFromMp3DecodeData(&data,af);
- cb->totalTime = data.totalTime;
+ initAudioFormatFromMp3DecodeData(&data, &(dc->audioFormat));
+ getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+
+ dc->totalTime = data.totalTime;
dc->start = 0;
dc->state = DECODE_STATE_DECODE;
diff --git a/src/mp3_decode.h b/src/mp3_decode.h
index 06525e8c3..c50df55c1 100644
--- a/src/mp3_decode.h
+++ b/src/mp3_decode.h
@@ -28,7 +28,7 @@
/* this is primarily used in tag.c */
int getMp3TotalTime(char * file);
-int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int mp3_decode(OutputBuffer * cb, DecoderControl * dc);
#endif
diff --git a/src/mp4_decode.c b/src/mp4_decode.c
index 7f0825ff0..b5c006f9e 100644
--- a/src/mp4_decode.c
+++ b/src/mp4_decode.c
@@ -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_callback_t * mp4cb;
int32_t track;
@@ -152,7 +152,7 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
#endif
faacDecSetConfiguration(decoder,config);
- af->bits = 16;
+ dc->audioFormat.bits = 16;
mp4Buffer = NULL;
mp4BufferSize = 0;
@@ -169,8 +169,8 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
return -1;
}
- af->sampleRate = sampleRate;
- af->channels = channels;
+ dc->audioFormat.sampleRate = sampleRate;
+ dc->audioFormat.channels = channels;
time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
scale = mp4ff_time_scale(mp4fh,track);
@@ -184,7 +184,7 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
free(mp4cb);
return -1;
}
- cb->totalTime = ((float)time)/scale;
+ dc->totalTime = ((float)time)/scale;
numSamples = mp4ff_num_samples(mp4fh,track);
@@ -255,8 +255,10 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
#endif
- af->sampleRate = scale;
- af->channels = frameInfo.channels;
+ dc->audioFormat.sampleRate = scale;
+ dc->audioFormat.channels = frameInfo.channels;
+ getOutputAudioFormat(&(dc->audioFormat),
+ &(cb->audioFormat));
dc->state = DECODE_STATE_DECODE;
dc->start = 0;
}
diff --git a/src/mp4_decode.h b/src/mp4_decode.h
index 8116758db..31b266f16 100644
--- a/src/mp4_decode.h
+++ b/src/mp4_decode.h
@@ -29,7 +29,7 @@
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 length);
diff --git a/src/ogg_decode.c b/src/ogg_decode.c
index 202fd6d4b..2869be7cb 100644
--- a/src/ogg_decode.c
+++ b/src/ogg_decode.c
@@ -142,7 +142,7 @@ float ogg_getReplayGainScale(char ** comments) {
return 1.0;
}
-int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
+int ogg_decode(OutputBuffer * cb, DecoderControl * dc)
{
OggVorbis_File vf;
ov_callbacks callbacks;
@@ -167,12 +167,13 @@ int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
{
vorbis_info *vi=ov_info(&vf,-1);
- af->bits = 16;
- af->channels = vi->channels;
- af->sampleRate = vi->rate;
+ dc->audioFormat.bits = 16;
+ dc->audioFormat.channels = vi->channels;
+ 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->start = 0;
@@ -203,7 +204,8 @@ int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
if((test = ov_bitrate_instant(&vf))>0) {
bitRate = test/1000;
}
- doReplayGain(chunk,ret,af,replayGainScale);
+ doReplayGain(chunk,ret,&(dc->audioFormat),
+ replayGainScale);
sendDataToOutputBuffer(cb,dc,chunk,ret,
ov_time_tell(&vf),bitRate);
if(dc->stop) break;
diff --git a/src/ogg_decode.h b/src/ogg_decode.h
index 6710c5b8e..16928eac6 100644
--- a/src/ogg_decode.h
+++ b/src/ogg_decode.h
@@ -25,7 +25,7 @@
#include <stdio.h>
-int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int ogg_decode(OutputBuffer * cb, DecoderControl * dc);
int getOggTotalTime(char * file);
diff --git a/src/outputBuffer.h b/src/outputBuffer.h
index 00ff6b5bd..c00e8779b 100644
--- a/src/outputBuffer.h
+++ b/src/outputBuffer.h
@@ -21,6 +21,7 @@
#include "mpd_types.h"
#include "decode.h"
+#include "audio.h"
#define OUTPUT_BUFFER_DC_STOP -1
#define OUTPUT_BUFFER_DC_SEEK -2
@@ -34,7 +35,7 @@ typedef struct _OutputBuffer {
mpd_sint16 volatile end;
mpd_sint16 volatile next;
mpd_sint8 volatile wrap;
- float totalTime;
+ AudioFormat audioFormat;
} OutputBuffer;
void flushOutputBuffer(OutputBuffer * cb);
diff --git a/src/pcm_utils.h b/src/pcm_utils.h
index 109981274..839005ccf 100644
--- a/src/pcm_utils.h
+++ b/src/pcm_utils.h
@@ -33,5 +33,8 @@ void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
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
/* vim:set shiftwidth=4 tabstop=8 expandtab: */
diff --git a/src/playerData.h b/src/playerData.h
index 387eb07cf..875fa0817 100644
--- a/src/playerData.h
+++ b/src/playerData.h
@@ -35,7 +35,6 @@ extern int buffered_chunks;
typedef struct _PlayerData {
OutputBuffer buffer;
- AudioFormat audioFormat;
PlayerControl playerControl;
DecoderControl decoderControl;
} PlayerData;
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-04 20:26:31 +0000