aac decoding!
need to work on seeking for AAC also, don't reset cb->begin on seek to 0, instead just set cb->end=cb->begin, works much better for disabling seeking (like ADIF AAC's) git-svn-id: https://svn.musicpd.org/mpd/trunk@356 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes
parent
4c1eb9225d
commit
145f0b62d2
151
src/aac_decode.c
151
src/aac_decode.c
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@ -2,8 +2,6 @@
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* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
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* This project's homepage is: http://www.musicpd.org
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*
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* libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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@ -196,7 +194,11 @@ int initAacBuffer(char * file, AacBuffer * b, float * length) {
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if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate;
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}
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if(*length<0) return -1;
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if(*length<0) {
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fclose(b->infile);
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if(b->buffer) free(b->buffer);
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return -1;
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}
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return 0;
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}
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@ -215,59 +217,29 @@ int getAacTotalTime(char * file) {
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int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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/*FILE * fh;
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mp4ff_t * mp4fh;
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mp4ff_callback_t * mp4cb;
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int32_t track;
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float time;
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int32_t scale;
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float totalTime;
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faacDecHandle decoder;
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faacDecFrameInfo frameInfo;
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faacDecConfigurationPtr config;
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unsigned char * mp4Buffer;
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int mp4BufferSize;
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size_t bread;
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unsigned long sampleRate;
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unsigned char channels;
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long sampleId;
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long numSamples;
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int eof = 0;
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long dur;
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unsigned int sampleCount;
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char * sampleBuffer;
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size_t sampleBufferLen;
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unsigned int initial = 1;
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int chunkLen = 0;
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float * seekTable;
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/*float * seekTable;
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long seekTableEnd = -1;
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int seekPositionFound = 0;
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long offset;
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int seekPositionFound = 0;*/
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mpd_uint16 bitRate = 0;
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AacBuffer b;
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fh = fopen(dc->file,"r");
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if(!fh) {
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ERROR("failed to open %s\n",dc->file);
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return -1;
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}
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printf("aac_decode!\n");
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mp4cb = malloc(sizeof(mp4ff_callback_t));
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mp4cb->read = mp4_readCallback;
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mp4cb->seek = mp4_seekCallback;
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mp4cb->user_data = fh;
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mp4fh = mp4ff_open_read(mp4cb);
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if(!mp4fh) {
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ERROR("Input does not appear to be a mp4 stream.\n");
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free(mp4cb);
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fclose(fh);
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return -1;
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}
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track = mp4_getAACTrack(mp4fh);
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if(track < 0) {
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ERROR("No AAC track found in mp4 stream.\n");
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mp4ff_close(mp4fh);
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fclose(fh);
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free(mp4cb);
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if(initAacBuffer(dc->file,&b,&totalTime) < 0) {
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ERROR("Not AAC file no ADTS or ADIF headers found.\n");
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return -1;
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}
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@ -285,48 +257,33 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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af->bits = 16;
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mp4Buffer = NULL;
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mp4BufferSize = 0;
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mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
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if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
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< 0)
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fillAacBuffer(&b);
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if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
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&sampleRate,&channels)) < 0)
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{
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ERROR("Error initializing AAC decoder library.\n");
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ERROR("Error not a AAC stream.\n");
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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free(mp4cb);
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fclose(fh);
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fclose(b.infile);
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if(b.buffer) free(b.buffer);
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return -1;
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}
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af->sampleRate = sampleRate;
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af->channels = channels;
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time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
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scale = mp4ff_time_scale(mp4fh,track);
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if(mp4Buffer) free(mp4Buffer);
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if(scale < 0) {
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ERROR("Error getting audio format of mp4 AAC track.\n");
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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fclose(fh);
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free(mp4cb);
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return -1;
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}
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cb->totalTime = ((float)time)/scale;
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numSamples = mp4ff_num_samples(mp4fh,track);
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cb->totalTime = totalTime+0.5;
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dc->state = DECODE_STATE_DECODE;
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dc->start = 0;
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time = 0.0;
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seekTable = malloc(sizeof(float)*numSamples);
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advanceAacBuffer(&b,bread);
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fillAacBuffer(&b);
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for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
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if(dc->seek && seekTableEnd>1 &&
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/*seekTable = malloc(sizeof(float)*numSamples);*/
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do {
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/*if(dc->seek && seekTableEnd>1 &&
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seekTable[seekTableEnd]>=dc->seekWhere)
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{
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int i = 2;
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@ -335,9 +292,6 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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time = seekTable[sampleId];
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}
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dur = mp4ff_get_sample_duration(mp4fh,track,sampleId);
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offset = mp4ff_get_sample_offset(mp4fh,track,sampleId);
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if(sampleId>seekTableEnd) {
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seekTable[sampleId] = time;
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seekTableEnd = sampleId;
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@ -358,42 +312,36 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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dc->seek = 0;
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}
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if(dc->seek) continue;
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if(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
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&mp4BufferSize) == 0)
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{
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eof = 1;
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continue;
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}
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if(dc->seek) continue;*/
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if(dc->seek) {
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/*chunkLen = 0;
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cb->wrap = 0;
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cb->end = 0;*/
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dc->seekError = 1;
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dc->seek = 0;
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}
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sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer,
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b.bytesIntoBuffer);
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advanceAacBuffer(&b,frameInfo.bytesconsumed);
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sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
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mp4BufferSize);
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if(mp4Buffer) free(mp4Buffer);
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if(frameInfo.error > 0) {
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eof = 1;
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break;
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}
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if(channels*(dur+offset) > frameInfo.samples) {
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dur = frameInfo.samples;
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offset = 0;
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}
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sampleCount = (unsigned long)(dur*channels);
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sampleCount = (unsigned long)(frameInfo.samples);
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if(sampleCount>0) {
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initial =0;
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bitRate = frameInfo.bytesconsumed*8.0*
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frameInfo.channels*scale/
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frameInfo.channels*sampleRate/
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frameInfo.samples/1024+0.5;
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time+= (float)(frameInfo.samples)/channels/sampleRate;
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}
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sampleBufferLen = sampleCount*2;
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sampleBuffer+=offset*channels*2;
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while(sampleBufferLen>0 && !dc->seek) {
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size_t size = sampleBufferLen>CHUNK_SIZE-chunkLen ?
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CHUNK_SIZE-chunkLen:
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@ -427,7 +375,11 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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}
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}
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}
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}
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fillAacBuffer(&b);
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if(b.bytesIntoBuffer==0) eof = 1;
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} while (!eof);
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if(!dc->stop && !dc->seek && chunkLen>0) {
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cb->chunkSize[cb->end] = chunkLen;
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@ -440,11 +392,10 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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chunkLen = 0;
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}
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free(seekTable);
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/*free(seekTable);*/
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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fclose(fh);
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free(mp4cb);
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fclose(b.infile);
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if(b.buffer) free(b.buffer);
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if(dc->seek) dc->seek = 0;
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@ -452,7 +403,7 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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dc->state = DECODE_STATE_STOP;
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dc->stop = 0;
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}
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else dc->state = DECODE_STATE_STOP;*/
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else dc->state = DECODE_STATE_STOP;
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return 0;
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}
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@ -2,8 +2,6 @@
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* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
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* This project's homepage is: http://www.musicpd.org
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*
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* libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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@ -86,7 +86,7 @@ int audiofile_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
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while(!eof) {
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if(dc->seek) {
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cb->end = 0;
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cb->end = cb->begin;
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cb->wrap = 0;
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current = dc->seekWhere * af->sampleRate;
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afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);
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@ -40,6 +40,7 @@
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#endif
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#ifdef HAVE_FAAD
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#include "mp4_decode.h"
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#include "aac_decode.h"
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#endif
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#include <signal.h>
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@ -159,11 +160,11 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
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pc->totalTime-0.1 :
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pc->seekWhere;
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dc->seekWhere = 0 > dc->seekWhere ? 0 : dc->seekWhere;
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cb->begin = 0;
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dc->seekError = 0;
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dc->seek = 1;
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pc->elapsedTime = dc->seekWhere;
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pc->bitRate = 0;
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while(*decode_pid>0 && dc->seek) usleep(1000);
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if(dc->seekError) pc->elapsedTime = dc->seekWhere;
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}
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}
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pc->seek = 0;
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@ -229,6 +230,9 @@ int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af,
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break;
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#endif
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#ifdef HAVE_FAAD
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case DECODE_TYPE_AAC:
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dc->error = aac_decode(cb,af,dc);
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break;
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case DECODE_TYPE_MP4:
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dc->error = mp4_decode(cb,af,dc);
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break;
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13
src/decode.h
13
src/decode.h
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@ -21,6 +21,8 @@
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#include "../config.h"
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#include "mpd_types.h"
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#include <stdio.h>
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#include <sys/param.h>
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@ -38,11 +40,12 @@
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#define DECODE_ERROR_UNKTYPE 1
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typedef struct _DecoderControl {
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int state;
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int stop;
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int start;
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int error;
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int seek;
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mpd_sint8 state;
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mpd_sint8 stop;
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mpd_sint8 start;
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mpd_uint16 error;
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mpd_sint8 seek;
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mpd_sint8 seekError;
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double seekWhere;
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char file[MAXPATHLEN+1];
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} DecoderControl;
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@ -104,7 +104,7 @@ void flacPlayFile(char *file, Buffer * cb, AudioFormat * af,
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if(dc->seek) {
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FLAC__uint64 sampleToSeek = dc->seekWhere*
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af->sampleRate+0.5;
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cb->end = 0;
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cb->end = cb->begin;
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cb->wrap = 0;
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if(FLAC__file_decoder_seek_absolute(flacDec,
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sampleToSeek))
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@ -487,7 +487,7 @@ int mp3Read(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
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if(dc->seek) {
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long i = 0;
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cb->wrap = 0;
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cb->end = 0;
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cb->end = cb->begin;
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data->muteFrame = 1;
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while(i<data->highestFrame && dc->seekWhere >
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((float)mad_timer_count(data->times[i],
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@ -2,8 +2,6 @@
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* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
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* This project's homepage is: http://www.musicpd.org
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*
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* libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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@ -149,7 +147,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
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< 0)
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{
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ERROR("Error initializing AAC decoder library.\n");
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ERROR("Error not a AAC stream.\n");
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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free(mp4cb);
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@ -210,7 +208,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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if(dc->seek && seekPositionFound) {
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seekPositionFound = 0;
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chunkLen = 0;
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cb->end = 0;
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cb->end = cb->begin;
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cb->wrap = 0;
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dc->seek = 0;
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}
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@ -233,7 +231,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
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}
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if(channels*(dur+offset) > frameInfo.samples) {
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dur = frameInfo.samples;
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dur = frameInfo.samples/channels;
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offset = 0;
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}
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@ -2,8 +2,6 @@
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* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
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* This project's homepage is: http://www.musicpd.org
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*
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* libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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|
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@ -95,7 +95,7 @@ int ogg_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
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while(!eof) {
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if(dc->seek) {
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cb->end = 0;
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cb->end = cb->begin;
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cb->wrap = 0;
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chunkpos = 0;
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ov_time_seek_page(&vf,dc->seekWhere);
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