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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
*
* 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 "inputStream.h"
#include "mp4ff/mp4ff.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 */
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;
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
mp4AudioSpecificConfig mp4ASC;
#else
unsigned long dummy1_32;
unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
dummy7_8, dummy8_8;
#endif
mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff) {
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
#else
rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
&dummy3_8, &dummy4_8, &dummy5_8,
&dummy6_8, &dummy7_8, &dummy8_8);
#endif
free(buff);
if (rc < 0) continue;
return i;
}
}
/* can't decode this */
return -1;
}
uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer,
uint32_t length)
{
return readFromInputStream((InputStream*) inStream, buffer, 1, length);
}
uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position) {
return seekInputStream((InputStream *) inStream, position, SEEK_SET);
}
int mp4_decode(OutputBuffer * cb, DecoderControl * dc) {
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;
InputStream inStream;
if(openInputStream(&inStream,dc->file) < 0) {
ERROR("failed to open %s\n",dc->file);
return -1;
}
mp4cb = malloc(sizeof(mp4ff_callback_t));
mp4cb->read = mp4_inputStreamReadCallback;
mp4cb->seek = mp4_inputStreamSeekCallback;
mp4cb->user_data = &inStream;
mp4fh = mp4ff_open_read(mp4cb);
if(!mp4fh) {
ERROR("Input does not appear to be a mp4 stream.\n");
free(mp4cb);
closeInputStream(&inStream);
return -1;
}
track = mp4_getAACTrack(mp4fh);
if(track < 0) {
ERROR("No AAC track found in mp4 stream.\n");
mp4ff_close(mp4fh);
closeInputStream(&inStream);
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);
dc->audioFormat.bits = 16;
mp4Buffer = NULL;
mp4BufferSize = 0;
mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
< 0)
{
ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
free(mp4cb);
closeInputStream(&inStream);
return -1;
}
dc->audioFormat.sampleRate = sampleRate;
dc->audioFormat.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);
closeInputStream(&inStream);
free(mp4cb);
return -1;
}
dc->totalTime = ((float)time)/scale;
numSamples = mp4ff_num_samples(mp4fh,track);
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;
clearOutputBuffer(cb);
dc->seekChunk = cb->end;
dc->seek = 0;
}
if(dc->seek) continue;
if(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
&mp4BufferSize) == 0)
{
eof = 1;
continue;
}
#ifdef HAVE_FAAD_BUFLEN_FUNCS
sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
mp4BufferSize);
#else
sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer);
#endif
if(mp4Buffer) free(mp4Buffer);
if(frameInfo.error > 0) {
ERROR("error decoding MP4 file: %s\n",dc->file);
ERROR("faad2 error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
eof = 1;
break;
}
if(dc->state != DECODE_STATE_DECODE) {
channels = frameInfo.channels;
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
#endif
dc->audioFormat.sampleRate = scale;
dc->audioFormat.channels = frameInfo.channels;
getOutputAudioFormat(&(dc->audioFormat),
&(cb->audioFormat));
dc->state = DECODE_STATE_DECODE;
}
if(channels*(dur+offset) > frameInfo.samples) {
dur = frameInfo.samples/channels;
offset = 0;
}
sampleCount = (unsigned long)(dur*channels);
if(sampleCount>0) {
initial =0;
bitRate = frameInfo.bytesconsumed*8.0*
frameInfo.channels*scale/
frameInfo.samples/1000+0.5;
}
sampleBufferLen = sampleCount*2;
sampleBuffer+=offset*channels*2;
sendDataToOutputBuffer(cb,dc,sampleBuffer,
sampleBufferLen,time,bitRate);
if(dc->stop) {
eof = 1;
break;
}
}
flushOutputBuffer(cb);
free(seekTable);
faacDecClose(decoder);
mp4ff_close(mp4fh);
closeInputStream(&inStream);
free(mp4cb);
if(dc->state != DECODE_STATE_DECODE) return -1;
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;
}
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 */
/* vim:set shiftwidth=4 tabstop=8 expandtab: */
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