From 2ec1c5ff3c1ff67825fb449c9eab2c3e4ff441f6 Mon Sep 17 00:00:00 2001 From: Warren Dukes Date: Mon, 10 May 2004 12:35:18 +0000 Subject: some more work on organizing code for resampling/audioFormat conversion git-svn-id: https://svn.musicpd.org/mpd/trunk@968 09075e82-0dd4-0310-85a5-a0d7c8717e4f --- src/aac_decode.c | 12 ++++++----- src/aac_decode.h | 2 +- src/audiofile_decode.c | 27 ++++++++++++------------ src/audiofile_decode.h | 2 +- src/decode.c | 57 +++++++++++++++++++++++--------------------------- src/decode.h | 5 ++++- src/flac_decode.c | 29 +++++++++++++------------ src/flac_decode.h | 2 +- src/mp3_decode.c | 8 ++++--- src/mp3_decode.h | 2 +- src/mp4_decode.c | 16 +++++++------- src/mp4_decode.h | 2 +- src/ogg_decode.c | 14 +++++++------ src/ogg_decode.h | 2 +- src/outputBuffer.h | 3 ++- src/pcm_utils.h | 3 +++ src/playerData.h | 1 - 17 files changed, 100 insertions(+), 87 deletions(-) (limited to 'src') 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 #include @@ -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 #include @@ -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 -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 -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; -- cgit v1.2.3