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-rw-r--r--src/mp4_decode.c288
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diff --git a/src/mp4_decode.c b/src/mp4_decode.c
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+/* 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 "mp4_decode.h"
+
+#ifdef HAVE_FAAD
+
+#include "command.h"
+#include "utils.h"
+#include "audio.h"
+#include "log.h"
+#include "pcm_utils.h"
+
+#include "mp4ff/mp4ff.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <faad.h>
+
+int mp4_getAACTrack(mp4ff_t *infile) {
+ /* find AAC track */
+ int i, rc;
+ int numTracks = mp4ff_total_tracks(infile);
+
+ for (i = 0; i < numTracks; i++) {
+ unsigned char *buff = NULL;
+ int buff_size = 0;
+ mp4AudioSpecificConfig mp4ASC;
+
+ mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
+
+ if (buff) {
+ rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+ free(buff);
+ if (rc < 0) continue;
+ return i;
+ }
+ }
+
+ /* can't decode this */
+ return -1;
+}
+
+uint32_t mp4_readCallback(void *user_data, void *buffer, uint32_t length) {
+ return fread(buffer, 1, length, (FILE*)user_data);
+}
+
+uint32_t mp4_seekCallback(void *user_data, uint64_t position) {
+ return fseek((FILE*)user_data, position, SEEK_SET);
+}
+
+
+int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
+{
+ FILE * fh;
+ mp4ff_t * mp4fh;
+ mp4ff_callback_t * mp4cb;
+ int32_t track;
+ int32_t 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;
+
+ 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;
+ config->downMatrix = 1;
+ config->dontUpSampleImplicitSBR = 1;
+ 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);
+ 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(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;
+ {
+ 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;
+ }
+
+ 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:
+ 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;
+
+#ifdef WORDS_BIGENDIAN
+ pcm_changeBufferEndianness(sampleBuffer,
+ size,af->bits);
+#endif
+ memcpy(cb->chunks+cb->end*CHUNK_SIZE,
+ sampleBuffer,size);
+ cb->chunkSize[cb->end] = size;
+
+//#warning implement time for AAC
+ cb->times[cb->end] = 0;
+
+ ++cb->end;
+
+ if(cb->end>=buffered_chunks) {
+ cb->end = 0;
+ cb->wrap = 1;
+ }
+ }
+ }
+ }
+
+ if(dc->seek) dc->seek = 0;
+
+ if(dc->stop) {
+ dc->state = DECODE_STATE_STOP;
+ dc->stop = 0;
+ }
+ else dc->state = DECODE_STATE_STOP;
+ }
+
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+ fclose(fh);
+ free(mp4cb);
+
+ return 0;
+}
+
+#endif /* HAVE_FAAD */
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