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authorWarren Dukes <warren.dukes@gmail.com>2004-03-21 21:32:23 +0000
committerWarren Dukes <warren.dukes@gmail.com>2004-03-21 21:32:23 +0000
commit4c1eb9225d5a741e1234d48eb38a8df3da908259 (patch)
tree28d8821623d6f9cce26f316e18342f2c08250785 /src/aac_decode.c
parentb72f591641e9d311e813fb8e2ece643cc6562e56 (diff)
downloadmpd-4c1eb9225d5a741e1234d48eb38a8df3da908259.tar.gz
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add aac_decode.[ch] and start working on it
also, if locale is C or POSIX, set fs charset to iso-8859-1 git-svn-id: https://svn.musicpd.org/mpd/trunk@347 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/aac_decode.c')
-rw-r--r--src/aac_decode.c460
1 files changed, 460 insertions, 0 deletions
diff --git a/src/aac_decode.c b/src/aac_decode.c
new file mode 100644
index 000000000..26e430d06
--- /dev/null
+++ b/src/aac_decode.c
@@ -0,0 +1,460 @@
+/* the Music Player Daemon (MPD)
+ * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "aac_decode.h"
+
+#ifdef HAVE_FAAD
+
+#define AAC_MAX_CHANNELS 6
+
+#include "command.h"
+#include "utils.h"
+#include "audio.h"
+#include "log.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <faad.h>
+
+/* all code here is either based on or copied from FAAD2's frontend code */
+typedef struct {
+ long bytesIntoBuffer;
+ long bytesConsumed;
+ long fileOffset;
+ unsigned char *buffer;
+ int atEof;
+ FILE *infile;
+} AacBuffer;
+
+void fillAacBuffer(AacBuffer *b) {
+ if(b->bytesConsumed > 0) {
+ int bread;
+
+ if(b->bytesIntoBuffer) {
+ memmove((void *)b->buffer,(void*)(b->buffer+
+ b->bytesConsumed),b->bytesIntoBuffer);
+ }
+
+ if(!b->atEof) {
+ bread = fread((void *)(b->buffer+b->bytesIntoBuffer),1,
+ b->bytesConsumed,b->infile);
+ if(bread!=b->bytesConsumed) b->atEof = 1;
+ b->bytesIntoBuffer+=bread;
+ }
+
+ b->bytesConsumed = 0;
+
+ if(b->bytesIntoBuffer > 3) {
+ if(memcmp(b->buffer,"TAG",3)==0) b->bytesIntoBuffer = 0;
+ }
+ if(b->bytesIntoBuffer > 11) {
+ if(memcmp(b->buffer,"LYRICSBEGIN",11)==0) {
+ b->bytesIntoBuffer = 0;
+ }
+ }
+ if(b->bytesIntoBuffer > 8) {
+ if(memcmp(b->buffer,"APETAGEX",8)==0) {
+ b->bytesIntoBuffer = 0;
+ }
+ }
+ }
+}
+
+void advanceAacBuffer(AacBuffer * b, int bytes) {
+ b->fileOffset+=bytes;
+ b->bytesConsumed = bytes;
+ b->bytesIntoBuffer-=bytes;
+}
+
+static int adtsSampleRates[] = {96000,88200,64000,48000,44100,32000,24000,22050,
+ 16000,12000,11025,8000,7350,0,0,0};
+
+int adtsParse(AacBuffer * b, float * length) {
+ int frames, frameLength;
+ int tFrameLength = 0;
+ int sampleRate = 0;
+ float framesPerSec, bytesPerFrame;
+
+ /* Read all frames to ensure correct time and bitrate */
+ for(frames = 0; ;frames++) {
+ fillAacBuffer(b);
+
+ if(b->bytesIntoBuffer > 7) {
+ /* check syncword */
+ if (!((b->buffer[0] == 0xFF) &&
+ ((b->buffer[1] & 0xF6) == 0xF0)))
+ {
+ break;
+ }
+
+ if(frames==0) {
+ sampleRate = adtsSampleRates[
+ (b->buffer[2]&0x3c)>>2];
+ }
+
+ frameLength = ((((unsigned int)b->buffer[3] & 0x3))
+ << 11) | (((unsigned int)b->buffer[4])
+ << 3) | (b->buffer[5] >> 5);
+
+ tFrameLength+=frameLength;
+
+ if(frameLength > b->bytesIntoBuffer) break;
+
+ advanceAacBuffer(b,frameLength);
+ }
+ else break;
+ }
+
+ framesPerSec = (float)sampleRate/1024.0;
+ if(frames!=0) {
+ bytesPerFrame = (float)tFrameLength/(float)(frames*1000);
+ }
+ else bytesPerFrame = 0;
+ if(framesPerSec!=0) *length = (float)frames/framesPerSec;
+
+ return 1;
+}
+
+int initAacBuffer(char * file, AacBuffer * b, float * length) {
+ size_t fileread;
+ size_t bread;
+ size_t tagsize;
+
+ *length = -1;
+
+ memset(b,0,sizeof(AacBuffer));
+
+ b->infile = fopen(file,"r");
+ if(b->infile == NULL) return -1;
+
+ fseek(b->infile,0,SEEK_END);
+ fileread = ftell(b->infile);
+ fseek(b->infile,0,SEEK_SET);
+
+ b->buffer = malloc(FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
+ memset(b->buffer,0,FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
+
+ bread = fread(b->buffer,1,FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS,
+ b->infile);
+ b->bytesIntoBuffer = bread;
+ b->bytesConsumed = 0;
+ b->fileOffset = 0;
+
+ if(bread!=FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS) b->atEof = 1;
+
+ tagsize = 0;
+ if(!memcmp(b->buffer,"ID3",3)) {
+ tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
+ (b->buffer[8] << 7) | (b->buffer[9] << 0);
+
+ tagsize+=10;
+ advanceAacBuffer(b,tagsize);
+ fillAacBuffer(b);
+ }
+
+ if((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
+ adtsParse(b, length);
+ fseek(b->infile, tagsize, SEEK_SET);
+
+ bread = fread(b->buffer, 1,
+ FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS,
+ b->infile);
+ if(bread != FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS) b->atEof = 1;
+ else b->atEof = 0;
+ b->bytesIntoBuffer = bread;
+ b->bytesConsumed = 0;
+ b->fileOffset = tagsize;
+ }
+ else if(memcmp(b->buffer,"ADIF",4) == 0) {
+ int bitRate;
+ int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
+ bitRate = ((unsigned int)(b->buffer[4 + skipSize] & 0x0F)<<19) |
+ ((unsigned int)b->buffer[5 + skipSize]<<11) |
+ ((unsigned int)b->buffer[6 + skipSize]<<3) |
+ ((unsigned int)b->buffer[7 + skipSize] & 0xE0);
+
+ *length = fileread;
+ if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate;
+ }
+
+ if(*length<0) return -1;
+
+ return 0;
+}
+
+int getAacTotalTime(char * file) {
+ AacBuffer b;
+ float length;
+
+ if(initAacBuffer(file,&b,&length) < 0) return -1;
+
+ if(b.buffer) free(b.buffer);
+ fclose(b.infile);
+
+ return (int)(length+0.5);
+}
+
+
+int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
+ /*FILE * fh;
+ mp4ff_t * mp4fh;
+ mp4ff_callback_t * mp4cb;
+ int32_t track;
+ float time;
+ int32_t scale;
+ faacDecHandle decoder;
+ faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
+ unsigned char * mp4Buffer;
+ int mp4BufferSize;
+ unsigned long sampleRate;
+ unsigned char channels;
+ long sampleId;
+ long numSamples;
+ int eof = 0;
+ long dur;
+ unsigned int sampleCount;
+ char * sampleBuffer;
+ size_t sampleBufferLen;
+ unsigned int initial = 1;
+ int chunkLen = 0;
+ float * seekTable;
+ long seekTableEnd = -1;
+ int seekPositionFound = 0;
+ long offset;
+ mpd_uint16 bitRate = 0;
+
+ fh = fopen(dc->file,"r");
+ if(!fh) {
+ ERROR("failed to open %s\n",dc->file);
+ return -1;
+ }
+
+ mp4cb = malloc(sizeof(mp4ff_callback_t));
+ mp4cb->read = mp4_readCallback;
+ mp4cb->seek = mp4_seekCallback;
+ mp4cb->user_data = fh;
+
+ mp4fh = mp4ff_open_read(mp4cb);
+ if(!mp4fh) {
+ ERROR("Input does not appear to be a mp4 stream.\n");
+ free(mp4cb);
+ fclose(fh);
+ return -1;
+ }
+
+ track = mp4_getAACTrack(mp4fh);
+ if(track < 0) {
+ ERROR("No AAC track found in mp4 stream.\n");
+ mp4ff_close(mp4fh);
+ fclose(fh);
+ free(mp4cb);
+ return -1;
+ }
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder,config);
+
+ af->bits = 16;
+
+ mp4Buffer = NULL;
+ mp4BufferSize = 0;
+ mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
+
+ if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
+ < 0)
+ {
+ ERROR("Error initializing AAC decoder library.\n");
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ free(mp4cb);
+ fclose(fh);
+ return -1;
+ }
+
+ af->sampleRate = sampleRate;
+ af->channels = channels;
+ time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
+ scale = mp4ff_time_scale(mp4fh,track);
+
+ if(mp4Buffer) free(mp4Buffer);
+
+ if(scale < 0) {
+ ERROR("Error getting audio format of mp4 AAC track.\n");
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ fclose(fh);
+ free(mp4cb);
+ return -1;
+ }
+ cb->totalTime = ((float)time)/scale;
+
+ numSamples = mp4ff_num_samples(mp4fh,track);
+
+ dc->state = DECODE_STATE_DECODE;
+ dc->start = 0;
+ time = 0.0;
+
+ seekTable = malloc(sizeof(float)*numSamples);
+
+ for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
+ if(dc->seek && seekTableEnd>1 &&
+ seekTable[seekTableEnd]>=dc->seekWhere)
+ {
+ int i = 2;
+ while(seekTable[i]<dc->seekWhere) i++;
+ sampleId = i-1;
+ time = seekTable[sampleId];
+ }
+
+ dur = mp4ff_get_sample_duration(mp4fh,track,sampleId);
+ offset = mp4ff_get_sample_offset(mp4fh,track,sampleId);
+
+ if(sampleId>seekTableEnd) {
+ seekTable[sampleId] = time;
+ seekTableEnd = sampleId;
+ }
+
+ if(sampleId==0) dur = 0;
+ if(offset>dur) dur = 0;
+ else dur-=offset;
+ time+=((float)dur)/scale;
+
+ if(dc->seek && time>dc->seekWhere) seekPositionFound = 1;
+
+ if(dc->seek && seekPositionFound) {
+ seekPositionFound = 0;
+ chunkLen = 0;
+ cb->end = 0;
+ cb->wrap = 0;
+ dc->seek = 0;
+ }
+
+ if(dc->seek) continue;
+
+ if(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
+ &mp4BufferSize) == 0)
+ {
+ eof = 1;
+ continue;
+ }
+
+ sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
+ mp4BufferSize);
+ if(mp4Buffer) free(mp4Buffer);
+ if(frameInfo.error > 0) {
+ eof = 1;
+ break;
+ }
+
+ if(channels*(dur+offset) > frameInfo.samples) {
+ dur = frameInfo.samples;
+ offset = 0;
+ }
+
+ sampleCount = (unsigned long)(dur*channels);
+
+ if(sampleCount>0) {
+ initial =0;
+ bitRate = frameInfo.bytesconsumed*8.0*
+ frameInfo.channels*scale/
+ frameInfo.samples/1024+0.5;
+ }
+
+
+ sampleBufferLen = sampleCount*2;
+
+ sampleBuffer+=offset*channels*2;
+
+ while(sampleBufferLen>0 && !dc->seek) {
+ size_t size = sampleBufferLen>CHUNK_SIZE-chunkLen ?
+ CHUNK_SIZE-chunkLen:
+ sampleBufferLen;
+ while(cb->begin==cb->end && cb->wrap &&
+ !dc->stop && !dc->seek)
+ {
+ usleep(10000);
+ }
+ if(dc->stop) {
+ eof = 1;
+ break;
+ }
+ else if(!dc->seek) {
+ sampleBufferLen-=size;
+ memcpy(cb->chunks+cb->end*CHUNK_SIZE+chunkLen,
+ sampleBuffer,size);
+ cb->times[cb->end] = time;
+ cb->bitRate[cb->end] = bitRate;
+ sampleBuffer+=size;
+ chunkLen+=size;
+ if(chunkLen>=CHUNK_SIZE) {
+ cb->chunkSize[cb->end] = CHUNK_SIZE;
+ ++cb->end;
+
+ if(cb->end>=buffered_chunks) {
+ cb->end = 0;
+ cb->wrap = 1;
+ }
+ chunkLen = 0;
+ }
+ }
+ }
+ }
+
+ if(!dc->stop && !dc->seek && chunkLen>0) {
+ cb->chunkSize[cb->end] = chunkLen;
+ ++cb->end;
+
+ if(cb->end>=buffered_chunks) {
+ cb->end = 0;
+ cb->wrap = 1;
+ }
+ chunkLen = 0;
+ }
+
+ free(seekTable);
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ fclose(fh);
+ free(mp4cb);
+
+ if(dc->seek) dc->seek = 0;
+
+ if(dc->stop) {
+ dc->state = DECODE_STATE_STOP;
+ dc->stop = 0;
+ }
+ else dc->state = DECODE_STATE_STOP;*/
+
+ return 0;
+}
+
+#endif /* HAVE_FAAD */