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author | Warren Dukes <warren.dukes@gmail.com> | 2004-03-18 16:31:29 +0000 |
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committer | Warren Dukes <warren.dukes@gmail.com> | 2004-03-18 16:31:29 +0000 |
commit | ad94c1dcf3198180ba708b0b3598a4d982d90c87 (patch) | |
tree | a89394f3de028c5e0f5593244972ca15c7c5e40d | |
parent | f409d85bbdde60c3acc175c9ad30a6f9d372e9a8 (diff) | |
download | mpd-ad94c1dcf3198180ba708b0b3598a4d982d90c87.tar.gz mpd-ad94c1dcf3198180ba708b0b3598a4d982d90c87.tar.xz mpd-ad94c1dcf3198180ba708b0b3598a4d982d90c87.zip |
mp4/aac cleanups
git-svn-id: https://svn.musicpd.org/mpd/trunk@276 09075e82-0dd4-0310-85a5-a0d7c8717e4f
-rw-r--r-- | src/decode.c | 8 | ||||
-rw-r--r-- | src/libid3tag/config.h.in | 6 | ||||
-rw-r--r-- | src/libmad/config.h.in | 6 | ||||
-rw-r--r-- | src/mp4_decode.c | 173 |
4 files changed, 84 insertions, 109 deletions
diff --git a/src/decode.c b/src/decode.c index 6b0a2c71e..8cc1bf071 100644 --- a/src/decode.c +++ b/src/decode.c @@ -38,6 +38,9 @@ #ifdef HAVE_AUDIOFILE #include "audiofile_decode.h" #endif +#ifdef HAVE_FAAD +#include "mp4_decode.h" +#endif #include <signal.h> #include <sys/types.h> @@ -213,6 +216,11 @@ int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af, dc->error = mp3_decode(cb,af,dc); break; #endif +#ifdef HAVE_FAAD + case DECODE_TYPE_MP4: + dc->error = mp4_decode(cb,af,dc); + break; +#endif #ifdef HAVE_OGG case DECODE_TYPE_OGG: dc->error = ogg_decode(cb,af,dc); diff --git a/src/libid3tag/config.h.in b/src/libid3tag/config.h.in index b4f0f8997..ba35b4be9 100644 --- a/src/libid3tag/config.h.in +++ b/src/libid3tag/config.h.in @@ -72,6 +72,8 @@ /* Define to empty if `const' does not conform to ANSI C. */ #undef const -/* Define as `__inline' if that's what the C compiler calls it, or to nothing - if it is not supported. */ +/* Define to `__inline__' or `__inline' if that's what the C compiler + calls it, or to nothing if 'inline' is not supported under any name. */ +#ifndef __cplusplus #undef inline +#endif diff --git a/src/libmad/config.h.in b/src/libmad/config.h.in index 2a9671cd2..a29b58209 100644 --- a/src/libmad/config.h.in +++ b/src/libmad/config.h.in @@ -125,9 +125,11 @@ /* Define to empty if `const' does not conform to ANSI C. */ #undef const -/* Define as `__inline' if that's what the C compiler calls it, or to nothing - if it is not supported. */ +/* Define to `__inline__' or `__inline' if that's what the C compiler + calls it, or to nothing if 'inline' is not supported under any name. */ +#ifndef __cplusplus #undef inline +#endif /* Define to `int' if <sys/types.h> does not define. */ #undef pid_t diff --git a/src/mp4_decode.c b/src/mp4_decode.c index a96cc3536..8690dade5 100644 --- a/src/mp4_decode.c +++ b/src/mp4_decode.c @@ -36,6 +36,8 @@ #include <string.h> #include <faad.h> +/* all code here is either based on or copied from FAAD2's frontend code */ + int mp4_getAACTrack(mp4ff_t *infile) { /* find AAC track */ int i, rc; @@ -75,20 +77,25 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) mp4ff_t * mp4fh; mp4ff_callback_t * mp4cb; int32_t track; - int32_t time; + float time; int32_t scale; faacDecHandle decoder; faacDecFrameInfo frameInfo; faacDecConfigurationPtr config; - mp4AudioSpecificConfig mp4ASC; unsigned char * mp4Buffer; int mp4BufferSize; - unsigned int frameSize; - unsigned int useAacLength; unsigned long sampleRate; unsigned char channels; long sampleId; long numSamples; + int eof = 0; + int rc; + long dur; + unsigned int sampleCount; + char * sampleBuffer; + unsigned int initial = 1; + size_t sampleBufferLen; + fh = fopen(dc->file,"r"); if(!fh) { @@ -147,16 +154,8 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) af->channels = channels; time = mp4ff_get_track_duration_use_offsets(mp4fh,track); scale = mp4ff_time_scale(mp4fh,track); - frameSize = 1024; - useAacLength = 0; - if(mp4Buffer) { - if(AudioSpecificConfig(mp4Buffer,mp4BufferSize,&mp4ASC) >= 0) { - if(mp4ASC.frameLengthFlag==1) frameSize = 960; - if(mp4ASC.sbr_present_flag==1) frameSize*= 2; - } - free(mp4Buffer); - } + if(mp4Buffer) free(mp4Buffer); if(scale < 0) { ERROR("Error getting audio format of mp4 AAC track.\n"); @@ -172,110 +171,74 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) dc->state = DECODE_STATE_DECODE; dc->start = 0; - { - int eof = 0; - int rc; - long dur; - unsigned int sampleCount; - unsigned int delay = 0; - char * sampleBuffer; - unsigned int initial = 1; - size_t sampleBufferLen; - - for(sampleId=0; sampleId<numSamples && !eof; sampleId++) { - if(dc->seek) { - cb->end = 0; - cb->wrap = 0; -//#warning implement seeking here! - dc->seek = 0; - } + time = 0.0; + + for(sampleId=0; sampleId<numSamples && !eof; sampleId++) { + if(dc->seek) { + cb->end = 0; + cb->wrap = 0; +#warning implement seeking here! + dc->seek = 0; + } - dur = mp4ff_get_sample_duration(mp4fh,track,sampleId); - rc = mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer, - &mp4BufferSize); - - if(rc==0) eof = 1; - else { - sampleBuffer = faacDecDecode(decoder, - &frameInfo, - mp4Buffer, - mp4BufferSize); - if(mp4Buffer) free(mp4Buffer); - if(sampleId==0) dur = 0; - if(useAacLength || scale!=sampleRate) { - sampleCount = frameInfo.samples; - } - else { - sampleCount = (unsigned long)(dur * - frameInfo.channels); - if(!useAacLength && !initial && - (sampleId < numSamples/2) && - (sampleCount!= - frameInfo.samples)) - { - useAacLength = 1; - sampleCount = frameInfo.samples; - } - - if(initial && (sampleCount < frameSize* - frameInfo.channels) && - (frameInfo.samples > - sampleCount)) - { - delay = frameInfo.samples - - sampleCount; - } - - } - - if(sampleCount>0) initial =0; - sampleBufferLen = sampleCount*2; - sampleBuffer+=delay*2; - while(sampleBufferLen > 0) { - size_t size = sampleBufferLen> - CHUNK_SIZE? - CHUNK_SIZE: + dur = mp4ff_get_sample_duration(mp4fh,track,sampleId); + rc = mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer, + &mp4BufferSize); + + if(rc==0) { + eof = 1; + break; + } + + sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer, + mp4BufferSize); + if(mp4Buffer) free(mp4Buffer); + if(sampleId==0) dur = 0; + time+=((float)dur)/scale; + sampleCount = (unsigned long)(dur*channels); + + if(sampleCount>0) initial =0; + sampleBufferLen = sampleCount*2; + while(sampleBufferLen > 0) { + size_t size = sampleBufferLen>CHUNK_SIZE ? CHUNK_SIZE: sampleBufferLen; - while(cb->begin==cb->end && cb->wrap && - !dc->stop && !dc->seek) - { - usleep(10000); - } - if(dc->stop) { - eof = 1; - break; - } - else if(dc->seek) break; + while(cb->begin==cb->end && cb->wrap && + !dc->stop && !dc->seek) + { + usleep(10000); + } + if(dc->stop) { + eof = 1; + break; + } + else if(dc->seek) break; #ifdef WORDS_BIGENDIAN - pcm_changeBufferEndianness(sampleBuffer, - size,af->bits); + pcm_changeBufferEndianness(sampleBuffer,size,af->bits); #endif - memcpy(cb->chunks+cb->end*CHUNK_SIZE, - sampleBuffer,size); - cb->chunkSize[cb->end] = size; + sampleBufferLen-=size; + memcpy(cb->chunks+cb->end*CHUNK_SIZE,sampleBuffer,size); + cb->chunkSize[cb->end] = size; + sampleBuffer+=size; -//#warning implement time for AAC - cb->times[cb->end] = 0; - - ++cb->end; + cb->times[cb->end] = time; - if(cb->end>=buffered_chunks) { - cb->end = 0; - cb->wrap = 1; - } - } + ++cb->end; + + if(cb->end>=buffered_chunks) { + cb->end = 0; + cb->wrap = 1; } } + } - if(dc->seek) dc->seek = 0; + if(dc->seek) dc->seek = 0; - if(dc->stop) { - dc->state = DECODE_STATE_STOP; - dc->stop = 0; - } - else dc->state = DECODE_STATE_STOP; + if(dc->stop) { + dc->state = DECODE_STATE_STOP; + dc->stop = 0; } + else dc->state = DECODE_STATE_STOP; faacDecClose(decoder); mp4ff_close(mp4fh); |