int PlayThread_file()
{
int k;
if (buffercount > 0) {
for (k = 0; k < (768*2 - buffercount); k++)
buffer[k] = buffer[k + buffercount];
read_buffer_filestream(infile, buffer + (768*2) - buffercount, buffercount);
buffercount = 0;
}
sample_buffer = (char*)faacDecDecode(hDecoder, &frameInfo, buffer, 768*2);
if (frameInfo.error)
{
MessageBox(NULL, faacDecGetErrorMessage(frameInfo.error), "FAAD Error", MB_OK);
last_frame = 1;
}
buffercount += frameInfo.bytesconsumed;
bytecount += frameInfo.bytesconsumed;
if (bytecount >= fileread)
last_frame = 1;
return frameInfo.bytesconsumed;
}
void* NEAACDECAPI mtv_faacDecDecode(NeAACDecFrameInfo *hInfo,
uint8_t *buffer, uint32_t buffer_size)
{
unsigned char *pcm_o;
pcm_o = faacDecDecode(faac_hdec, &faac_finfo, buffer, buffer_size); //decode a audio packet
*hInfo = faac_finfo;
return (void *)pcm_o;
}
int AacPcm::processData(MediaInfo *infos, ChunkList *chunk_list, bool *killswitch)
{
DWORD BytesDecoded=0;
char *bufout=0;
ChunkInfosI *ci=0;
svc_fileReader *reader=0;
if(!FindBitrate && !chunk_list)
ERROR_processData("chunk_list==NULL"); // is this case possible?
if(!(reader=infos->getReader()))
ERROR_processData("File doesn\'t exists");
if(chunk_list)
{
if(!(ci=new ChunkInfosI()))
ERROR_processData("Memory allocation error: ci");
ci->addInfo("srate", Samplerate);
ci->addInfo("bps", bps);
ci->addInfo("nch", Channels);
}
if(!IsAAC) // MP4 file --------------------------------------------------------------------------
{
unsigned __int32 buffer_size=0;
int rc;
if(newpos_ms>-1)
{
MP4Duration duration=MP4ConvertToTrackDuration(mp4File,track,newpos_ms,MP4_MSECS_TIME_SCALE);
sampleId=MP4GetSampleIdFromTime(mp4File,track,duration,0);
bytes_read=(DWORD)(((float)newpos_ms*file_info.bitrate)/(8*1000));
reader->seek(bytes_read); // updates slider
newpos_ms=-1;
}
do
{
buffer=NULL;
if(sampleId>=numSamples)
ERROR_processData(0);
rc=MP4ReadSample(mp4File, track, sampleId++, (unsigned __int8 **)&buffer, &buffer_size, NULL, NULL, NULL, NULL);
if(rc==0 || buffer==NULL)
{
FREE_ARRAY(buffer);
ERROR_processData("MP4ReadSample")
}
bufout=(char *)faacDecDecode(hDecoder,&frameInfo,buffer,buffer_size);
BytesDecoded=frameInfo.samples*sizeof(short);
FREE_ARRAY(buffer);
// to update the slider
bytes_read+=buffer_size;
reader->seek(bytes_read);
}while(!BytesDecoded && !frameInfo.error);
unsigned int faad_decoder::update(void *dest, unsigned int const num_samples_to_write)
{
if (!is_initialized())
return 0;
unsigned int multiplier = playback_properties_.num_channels * get_sample_size(playback_properties_.sample_type_);
faacDecFrameInfo info;
while (out_buffer.size() < (num_samples_to_write * multiplier))
{
size_t actual_in_size = in_buffer.size();
if (actual_in_size < 32768)
{
size_t old_in_size = actual_in_size;
in_buffer.resize(old_in_size + 32768);
long num_read_bytes = source_->read(&in_buffer[old_in_size], 32768);
if (num_read_bytes <= 0)
break;
actual_in_size = old_in_size + num_read_bytes;
}
void *output_samples = faacDecDecode(*faad_handle, &info, reinterpret_cast < unsigned char* > (&in_buffer[0]), actual_in_size);
unsigned int num_remaining_bytes = actual_in_size;
num_remaining_bytes -= info.bytesconsumed;
std::memmove(&in_buffer[0], &in_buffer[info.bytesconsumed], num_remaining_bytes);
in_buffer.resize(num_remaining_bytes);
if ((output_samples == 0) || (info.error != 0) || (info.channels == 0))
{
initialized = false;
break;
}
if (info.samples > 0)
{
size_t old_out_size = out_buffer.size();
out_buffer.resize(old_out_size + info.samples * playback_properties_.num_channels);
std::memcpy(&out_buffer[old_out_size], output_samples, info.samples * playback_properties_.num_channels);
}
}
size_t num_bytes_to_copy = std::min(size_t(num_samples_to_write * multiplier), out_buffer.size());
std::memcpy(dest, &out_buffer[0], num_bytes_to_copy);
if (out_buffer.size() > num_bytes_to_copy)
std::memmove(&out_buffer[0], &out_buffer[num_bytes_to_copy], out_buffer.size() - num_bytes_to_copy);
out_buffer.resize(out_buffer.size() - num_bytes_to_copy);
current_position += num_bytes_to_copy / multiplier;
return num_bytes_to_copy / multiplier;
}
bool CCoreAACDecoder::Decode(BYTE *pSrc, DWORD SrcLength, BYTE *pDst, DWORD DstLength, DWORD *ActualDstLength)
{
faacDecFrameInfo frameInfo;
short *outsamples = (short *)faacDecDecode(m_decHandle, &frameInfo, pSrc, DstLength);
if (frameInfo.error)
{
NOTE2("CCoreAACDecoder::Decode Error %d [%s]\n",
frameInfo.error, faacDecGetErrorMessage(frameInfo.error));
return false;
}
m_brCalcFrames++;
m_DecodedFrames++;
m_brBytesConsumed += SrcLength;
if(m_brCalcFrames == 43)
{
m_Bitrate = (int)((m_brBytesConsumed * 8) / (m_DecodedFrames / 43.07));
m_brCalcFrames = 0;
}
if (!frameInfo.error && outsamples)
{
int channelidx = frameInfo.channels-1;
if(chmap[channelidx][0])
{
// dshow remapping
short *dstBuffer = (short*)pDst;
for(unsigned int i = 0;
i < frameInfo.samples;
i += frameInfo.channels, outsamples += frameInfo.channels)
{
for(unsigned int j=1; j <= frameInfo.channels; j++)
{
*dstBuffer++ = outsamples[chmap[channelidx][j]];
}
}
}
else
{
memcpy(pDst, outsamples, frameInfo.samples * sizeof(short));
}
}
else
return false;
*ActualDstLength = frameInfo.samples * sizeof(short);
return true;
}
int PlayThread_memmap()
{
sample_buffer = (char*)faacDecDecode(hDecoder, &frameInfo,
memmap_buffer + memmap_index, fileread - memmap_index - 1);
if (frameInfo.error)
{
MessageBox(NULL, faacDecGetErrorMessage(frameInfo.error), "FAAD Error", MB_OK);
last_frame = 1;
}
memmap_index += frameInfo.bytesconsumed;
if (memmap_index >= fileread)
last_frame = 1;
return frameInfo.bytesconsumed;
}
int fdpl_decode_AAC(int def_srate, const char *file_path, size_t &c, size_t &sr, size_t &samples, std::vector<float> * ch1, std::vector<float> *ch2)
{
int tagsize;
uint32_t samplerate;
unsigned char channels;
void *sample_buffer;
FILE *infile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
/* declare variables for buffering */
DEC_BUFF_VARS
infile = fopen(file_path, "rb");
if (infile == NULL)
{
/* unable to open file */
std::cerr<<"Error opening file"<<std::endl;
return 1;
}
INIT_BUFF(infile)
tagsize = id3v2_tag(buffer);//avoids ID3 tag (useful for mp3 as well)
if (tagsize)
{
UPDATE_BUFF_SKIP(tagsize)
}
hDecoder = faacDecOpen();
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
config = faacDecGetCurrentConfiguration(hDecoder);
if (def_srate)
config->defSampleRate = def_srate;
config->defObjectType = LC;//default
config->outputFormat = FAAD_FMT_FLOAT;
config -> downMatrix = 0; //set to 1
faacDecSetConfiguration(hDecoder, config);
if ((bytesconsumed = faacDecInit(hDecoder, buffer, bytes_in_buffer,
&samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
std::cerr<<"Error initializing decoder library."<<std::endl;
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 1;
}
buffer_index += bytesconsumed;
do
{
/* update buffer */
UPDATE_BUFF_READ
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer, bytes_in_buffer);
/* update buffer indices */
UPDATE_BUFF_IDX(frameInfo)
if (frameInfo.error > 0)
{
std::cerr<<"Error: %s\n"<<faacDecGetErrorMessage(frameInfo.error)<<std::endl;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0) && (frameInfo.channels < 3))
{
size_t samples_channel = (size_t)frameInfo.samples/(size_t)frameInfo.channels;
samples += samples_channel;
float *pcm;
pcm = (float *)sample_buffer;
for(size_t i=0; i<samples_channel; i++){
//writing the samples of each channel in separate float vectors
for(size_t j=0; j<frameInfo.channels; j++){
if(j==0){
ch1 -> push_back(pcm[i*frameInfo.channels+j]);
}else if(j==1){
ch2 -> push_back(pcm[i*frameInfo.channels+j]);
}else{
std::cerr<<"Unexpected number of channels: "<<j<<std::endl;
return -1;
}
}
}
}
if (buffer_index >= fileread)
sample_buffer = NULL; /* to make sure it stops now */
} while (sample_buffer != NULL);
c = frameInfo.channels;
sr = frameInfo.samplerate;
faacDecClose(hDecoder);
fclose(infile);
END_BUFF
return frameInfo.error;
}
static GF_Err FAAD_ProcessData(GF_MediaDecoder *ifcg,
char *inBuffer, u32 inBufferLength,
u16 ES_ID, u32 *CTS,
char *outBuffer, u32 *outBufferLength,
u8 PaddingBits, u32 mmlevel)
{
void *buffer;
unsigned short *conv_in, *conv_out;
u32 i, j;
FAADCTX();
/*check not using scalabilty*/
if (ctx->ES_ID != ES_ID) return GF_BAD_PARAM;
/*if late or seeking don't decode*/
switch (mmlevel) {
case GF_CODEC_LEVEL_SEEK:
// case GF_CODEC_LEVEL_DROP:
*outBufferLength = 0;
return GF_OK;
default:
break;
}
if (ctx->out_size > *outBufferLength) {
*outBufferLength = ctx->out_size;
return GF_BUFFER_TOO_SMALL;
}
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[FAAD] Decoding AU\n"));
buffer = faacDecDecode(ctx->codec, &ctx->info, (unsigned char *) inBuffer, inBufferLength);
if (ctx->info.error>0) {
GF_LOG(GF_LOG_WARNING, GF_LOG_CODEC, ("[FAAD] Error decoding AU %s\n", faacDecGetErrorMessage(ctx->info.error) ));
*outBufferLength = 0;
//reinit if error
FAAD_AttachStream((GF_BaseDecoder *)ifcg, ctx->esd);
return GF_NON_COMPLIANT_BITSTREAM;
}
if (!ctx->info.samples || !buffer || !ctx->info.bytesconsumed) {
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[FAAD] empty/non complete AU\n"));
*outBufferLength = 0;
return GF_OK;
}
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[FAAD] AU decoded\n"));
/*FAAD froces us to decode a frame to get channel cfg*/
if (ctx->signal_mc) {
s32 ch, idx;
ctx->signal_mc = GF_FALSE;
idx = 0;
/*NOW WATCH OUT!! FAAD may very well decide to output more channels than indicated!!!*/
ctx->num_channels = ctx->info.channels;
/*get cfg*/
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_FRONT_LEFT);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_FRONT_RIGHT);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_FRONT_CENTER);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_LFE);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_BACK_LEFT);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_BACK_RIGHT);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_BACK_CENTER);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_SIDE_LEFT);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
ch = FAAD_GetChannelPos(ctx, GF_AUDIO_CH_SIDE_RIGHT);
if (ch>=0) {
ctx->ch_reorder[idx] = ch;
idx++;
}
*outBufferLength = ctx->out_size;
if (sizeof(short) * ctx->info.samples > *outBufferLength) {
*outBufferLength = ctx->out_size = (u32) (sizeof(short)*ctx->info.samples);
}
return GF_BUFFER_TOO_SMALL;
}
if (sizeof(short) * ctx->info.samples > *outBufferLength) {
*outBufferLength = (u32) (sizeof(short)*ctx->info.samples);
return GF_BUFFER_TOO_SMALL;
}
/*we assume left/right order*/
if (ctx->num_channels<=2) {
memcpy(outBuffer, buffer, sizeof(short)* ctx->info.samples);
*outBufferLength = (u32) (sizeof(short)*ctx->info.samples);
return GF_OK;
}
conv_in = (unsigned short *) buffer;
conv_out = (unsigned short *) outBuffer;
for (i=0; i<ctx->info.samples; i+=ctx->info.channels) {
for (j=0; j<ctx->info.channels; j++) {
conv_out[i + j] = conv_in[i + ctx->ch_reorder[j]];
}
}
*outBufferLength = (u32) (sizeof(short)*ctx->info.samples);
return GF_OK;
}
int main(int argc, char *argv[])
{
int k;
short sample_buffer[1024*MAX_CHANNELS];
int writeToStdio = 0, out_dir_set = 0;
int def_sr_set = 0, use_ltp = 0;
int def_srate;
int showHelp = 0;
char *fnp;
int result;
int first_time = 1;
char out_dir[255];
char aacFileName[255];
char audioFileName[255];
FILE *sndfile;
FILE *infile;
faacDecHandle hDecoder;
faacDecConfigurationPtr config;
char percent[200];
long buffercount;
unsigned long bytesconsumed, samples, fileread, bytecount;
unsigned char *buffer;
unsigned long samplerate, channels, tagsize;
/* System dependant types */
#ifdef _WIN32
long begin, end;
#else
clock_t begin;
#endif
fprintf(stderr, "FAAD (Freeware AAC Decoder) Compiled on: " __DATE__ "\n");
fprintf(stderr, "FAAD homepage: %s\n", "http://www.audiocoding.com");
/* begin process command line */
progName = argv[0];
while (1) {
int c = -1;
int option_index = 0;
static struct option long_options[] = {
{ "outputdir", 0, 0, 'o' },
{ "samplerate", 0, 0, 's' },
{ "ltp", 0, 0, 'l' },
{ "stdio", 0, 0, 'w' },
{ "help", 0, 0, 'h' }
};
c = getopt_long(argc, argv, "o:s:whl",
long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 'o': {
if (optarg) {
char dr[255];
if (sscanf(optarg, "%s", dr) < 1) {
out_dir_set = 0;
} else {
out_dir_set = 1;
strcpy(out_dir, dr);
}
} else {
out_dir_set = 0;
}
break;
}
case 's': {
if (optarg) {
char dr[255];
if (sscanf(optarg, "%s", dr) < 1) {
def_sr_set = 0;
} else {
def_sr_set = 1;
def_srate = atoi(dr);
}
} else {
out_dir_set = 0;
}
break;
}
case 'w': {
writeToStdio = 1;
break;
}
case 'l': {
use_ltp = 1;
break;
}
case 'h': {
showHelp = 1;
break;
}
default:
break;
}
}
/* check that we have at least two non-option arguments */
/* Print help if requested */
if (((argc - optind) < 1) || showHelp)
{
usage();
return 1;
}
/* point to the specified file name */
strcpy(aacFileName, argv[optind]);
#ifdef _WIN32
begin = GetTickCount();
#else
begin = clock();
#endif
buffer = (unsigned char*)malloc(768*MAX_CHANNELS);
memset(buffer, 0, 768*MAX_CHANNELS);
infile = fopen(aacFileName, "rb");
if (infile == NULL)
{
/* unable to open file */
fprintf(stderr, "Error opening file: %s\n", aacFileName);
return 1;
}
fseek(infile, 0, SEEK_END);
fileread = ftell(infile);
fseek(infile, 0, SEEK_SET);
buffercount = bytecount = 0;
fread(buffer, 1, 768*MAX_CHANNELS, infile);
tagsize = id3v2_tag(buffer);
if (tagsize) {
fseek(infile, tagsize, SEEK_SET);
bytecount = tagsize;
buffercount = 0;
fread(buffer, 1, 768*MAX_CHANNELS, infile);
}
hDecoder = faacDecOpen();
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
config = faacDecGetCurrentConfiguration(hDecoder);
if (def_sr_set)
config->defSampleRate = def_srate;
if (use_ltp)
config->defObjectType = LTP;
faacDecSetConfiguration(hDecoder, config);
if((buffercount = faacDecInit(hDecoder, buffer, &samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
fprintf(stderr, "Error initializing decoder library.\n");
return 1;
}
if (buffercount > 0) {
bytecount += buffercount;
for (k = 0; k < (768*MAX_CHANNELS - buffercount); k++)
buffer[k] = buffer[k + buffercount];
fread(buffer + (768*MAX_CHANNELS) - buffercount, 1, buffercount, infile);
buffercount = 0;
}
fprintf(stderr, "Busy decoding %s", aacFileName);
if (tagsize)
fprintf(stderr, " (has ID3v2)\n");
else
fprintf(stderr, "\n");
/* Only calculate the path and open the file for writing if we are not writing to stdout. */
if(!writeToStdio)
{
if (out_dir_set)
combine_path(audioFileName, out_dir, aacFileName);
else
strcpy(audioFileName, aacFileName);
fnp = (char *)strrchr(audioFileName,'.');
if (fnp)
fnp[0] = '\0';
strcat(audioFileName, ".wav");
}
first_time = 1;
do
{
if (buffercount > 0) {
for (k = 0; k < (768*MAX_CHANNELS - buffercount); k++)
buffer[k] = buffer[k + buffercount];
fread(buffer + (768*MAX_CHANNELS) - buffercount, 1, buffercount, infile);
buffercount = 0;
}
result = faacDecDecode(hDecoder, buffer, &bytesconsumed, sample_buffer, &samples);
if (result == FAAD_FATAL_ERROR)
fprintf(stderr, "Fatal error decoding file\n");
if (result == FAAD_ERROR)
fprintf(stderr, "Error decoding frame\n");
buffercount += bytesconsumed;
bytecount += bytesconsumed;
if (bytecount >= fileread)
result = 9999; /* to make sure it stops now */
sprintf(percent, "%.2f decoding %s.",
min((double)(bytecount*100)/fileread,100), aacFileName);
fprintf(stderr, "%s\r", percent);
#ifdef _WIN32
SetConsoleTitle(percent);
#endif
if ((result == FAAD_OK) && (samples > 0))
{
if(first_time)
{
first_time = 0;
if(!writeToStdio)
{
sndfile = fopen(audioFileName, "w+b");
if (sndfile==NULL)
{
fprintf(stderr, "Unable to create the file << %s >>.\n", audioFileName);
continue;
}
CreateWavHeader(sndfile, channels, samplerate, 16);
}
else
{
sndfile = stdout;
}
}
fwrite(sample_buffer, sizeof(short), 1024*channels, sndfile);
}
} while (result <= FAAD_OK_CHUPDATE);
faacDecClose(hDecoder);
fclose(infile);
if(!writeToStdio)
UpdateWavHeader(sndfile);
/* Only close if we are not writing to stdout */
if(!writeToStdio)
if(sndfile)
fclose(sndfile);
#ifdef _WIN32
end = GetTickCount();
fprintf(stderr, "Decoding %s took: %d sec.\n", aacFileName, (end-begin)/1000);
SetConsoleTitle("FAAD");
#else
/* clock() grabs time since the start of the app but when we decode multiple files,
each file has its own starttime (begin). */
fprintf(stderr, "Decoding %s took: %5.2f sec.\n", aacFileName,
(float)(clock() - begin)/(float)CLOCKS_PER_SEC);
#endif
if (buffer) free(buffer);
return 0;
}
static gint
xmms_faad_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len, xmms_error_t *err)
{
xmms_faad_data_t *data;
xmms_error_t error;
faacDecFrameInfo frameInfo;
gpointer sample_buffer;
guint size, bytes_read = 0;
data = xmms_xform_private_data_get (xform);
g_return_val_if_fail (data, -1);
size = MIN (data->outbuf->len, len);
while (size == 0) {
gboolean need_read;
/* MP4 demuxer always gives full packets so we need different handling */
if (data->filetype == FAAD_TYPE_MP4)
need_read = (data->buffer_length == 0);
else
need_read = (data->buffer_length < data->buffer_size);
if (need_read) {
bytes_read = xmms_xform_read (xform,
(gchar *) data->buffer + data->buffer_length,
data->buffer_size - data->buffer_length,
&error);
if (bytes_read <= 0 && data->buffer_length == 0) {
XMMS_DBG ("EOF");
return 0;
}
data->buffer_length += bytes_read;
}
sample_buffer = faacDecDecode (data->decoder, &frameInfo, data->buffer,
data->buffer_length);
g_memmove (data->buffer, data->buffer + frameInfo.bytesconsumed,
data->buffer_length - frameInfo.bytesconsumed);
data->buffer_length -= frameInfo.bytesconsumed;
bytes_read = frameInfo.samples * xmms_sample_size_get (data->sampleformat);
if (bytes_read > 0 && frameInfo.error == 0) {
gint32 temp, toskip = 0;
if (data->samplerate != frameInfo.samplerate ||
data->channels != frameInfo.channels) {
/* We should inform output to change parameters somehow */
XMMS_DBG ("Output format changed in the middle of a read!");
data->samplerate = frameInfo.samplerate;
data->channels = frameInfo.channels;
}
if (xmms_xform_auxdata_get_int (xform, "frame_offset", &temp)) {
toskip = (temp * frameInfo.channels *
xmms_sample_size_get (data->sampleformat));
}
if (xmms_xform_auxdata_get_int (xform, "frame_duration", &temp)) {
bytes_read = (temp * frameInfo.channels *
xmms_sample_size_get (data->sampleformat));
}
g_string_append_len (data->outbuf, sample_buffer + toskip,
bytes_read - toskip);
} else if (frameInfo.error > 0) {
XMMS_DBG ("ERROR %d in faad decoding: %s", frameInfo.error,
faacDecGetErrorMessage (frameInfo.error));
return -1;
}
size = MIN (data->outbuf->len, len);
}
memcpy (buf, data->outbuf->str, size);
g_string_erase (data->outbuf, 0, size);
return size;
}
uint8_t ADM_faad::run(uint8_t *inptr, uint32_t nbIn, float *outptr, uint32_t *nbOut)
{
uint32_t xin;
long int res;
void *outbuf;
faacDecFrameInfo info;
int max=0;
#ifdef OLD_FAAD_PROTO
unsigned long int srate;
#else
uint32_t srate;
#endif
unsigned char chan=0;
uint8_t first=0;
ADM_assert(_instance);
*nbOut=0;
if(!_inited) // we still have'nt initialized the codec
{
// Try
printf("Trying with %d bytes\n",nbIn);
res=faacDecInit(_instance,inptr,nbIn,&srate,&chan);
if(res>=0)
{
printf("Faad Inited : rate:%d chan:%d off:%ld\n",srate,chan,res);
_inited=1;
first=1;
_inbuffer=0;
inptr+=res;
nbIn-=res;
}
}
if(!_inited)
{
printf("No dice...\n");
return 1;
}
// The codec is initialized, feed him
do
{
max=FAAD_BUFFER-_inbuffer;
if(nbIn<max) max=nbIn;
memcpy(_buffer+_inbuffer,inptr,max);
inptr+=max;
nbIn-=max;
_inbuffer+=max;
/*
if(_inbuffer<FAAD_MIN_STREAMSIZE*2)
{
return 1;
}*/
memset(&info,0,sizeof(info));
//printf("%x %x\n",_buffer[0],_buffer[1]);
outbuf = faacDecDecode(_instance, &info, _buffer, _inbuffer);
if(info.error)
{
printf("Faad: Error %d :%s\n",info.error,
faacDecGetErrorMessage(info.error));
printf("Bye consumed %u, bytes dropped %u \n",info.bytesconsumed,_inbuffer);
_inbuffer=0; // Purge buffer
return 1;
}
if(first)
{
printf("Channels : %d\n",info.channels);
printf("Frequency: %d\n",info.samplerate);
printf("SBR : %d\n",info.sbr);
}
xin=info.bytesconsumed ;
if(xin>_inbuffer) xin=0;
memmove(_buffer,_buffer+xin,_inbuffer-xin);
_inbuffer-=xin;
if(info.samples)
{
*nbOut+= info.samples;
int16_t *in = (int16_t *) outbuf;
for (int i = 0; i < info.samples; i++) {
*(outptr++) = (float)*in / 32768;
in++;
}
}
}while(nbIn);
return 1;
}
int fdpl_decode_MP4(const char *file_path, size_t &ch, size_t &sr, size_t &samples, std::vector<float> *ch1, std::vector<float> *ch2)
{
unsigned int track;
uint32_t samplerate;//these for init, we prefer to take the ones at frameInfo
unsigned char channels;
void *sample_buffer;
mp4ff_t *infile;
FILE *mp4File;
int sampleId, numSamples;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
unsigned char *buffer;
unsigned int buffer_size;
/* initialise the callback structure */
mp4ff_callback_t *mp4cb;
mp4cb = (mp4ff_callback_t *)malloc(sizeof(mp4ff_callback_t));
mp4File = fopen(file_path, "rb");
mp4cb->read = read_callback;
mp4cb->seek = seek_callback;
mp4cb->user_data = mp4File;
infile = mp4ff_open_read(mp4cb);
if (!infile)
{
/* unable to open file */
std::cerr<<"Error opening file "<< file_path <<std::endl;
return 1;
}
if ((track = GetAACTrack(infile)) < 0)
{
std::cerr<<"Unable to find correct AAC sound track in the MP4 file"<<std::endl;
mp4ff_close(infile);
free(mp4cb);
fclose(mp4File);
return 1;
}
buffer = NULL;
buffer_size = 0;
mp4ff_get_decoder_config(infile, track, &buffer, &buffer_size);
hDecoder = faacDecOpen();
if(faacDecInit2(hDecoder, buffer, buffer_size, &samplerate, &channels) < 0)
{
/* If some error initializing occured, skip the file */
std::cerr<<"Error initializing decoder library"<<std::endl;
faacDecClose(hDecoder);
mp4ff_close(infile);
free(mp4cb);
fclose(mp4File);
return 1;
}
if (buffer)
free(buffer);
numSamples = mp4ff_num_samples(infile, track);
samples = 0;
//firstTime = 1;
for (sampleId = 0; sampleId < numSamples; sampleId++)
{
int rc;
/* get access unit from MP4 file */
buffer = NULL;
buffer_size = 0;
rc = mp4ff_read_sample(infile, track, sampleId, &buffer, &buffer_size);
if (rc == 0)
{
std::cerr<<"error while decoding"<<std::endl;
faacDecClose(hDecoder);
mp4ff_close(infile);
free(mp4cb);
fclose(mp4File);
return 1;
}
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer, buffer_size);
if (buffer)
free(buffer);
if ((frameInfo.error == 0) && (frameInfo.samples > 0) && (frameInfo.channels < 3))
{
size_t samples_channel = (size_t)frameInfo.samples/(size_t)frameInfo.channels;
samples += samples_channel;
float *pcm;
pcm = (float *)sample_buffer;
for(size_t i=0; i<samples_channel; i++){
//writing the samples of each channel in separate float vectors
for(size_t j=0; j<frameInfo.channels; j++){
if(j==0){
ch1 -> push_back(pcm[i*frameInfo.channels+j]);
}else if(j==1){
ch2 -> push_back(pcm[i*frameInfo.channels+j]);
}else{
std::cerr<<"Unexpected number of channels: "<<j<<std::endl;
return -1;
}
}
}
}
if (frameInfo.error > 0)
{
std::cerr<<"error: "<<faacDecGetErrorMessage(frameInfo.error)<<std::endl;
break;
}
}
ch = frameInfo.channels;
sr = frameInfo.samplerate;
faacDecClose(hDecoder);
mp4ff_close(infile);
free(mp4cb);
fclose(mp4File);
return frameInfo.error;
}
/*
* Decode task call for FAAC
*/
static int aac_decode (codec_data_t *ptr,
uint64_t ts,
int from_rtp,
int *sync_frame,
uint8_t *buffer,
uint32_t buflen,
void *userdata)
{
aac_codec_t *aac = (aac_codec_t *)ptr;
unsigned long bytes_consummed;
int bits = -1;
// struct timezone tz;
if (aac->m_record_sync_time) {
aac->m_current_frame = 0;
aac->m_record_sync_time = 0;
aac->m_current_time = ts;
aac->m_last_rtp_ts = ts;
} else {
if (aac->m_last_rtp_ts == ts) {
aac->m_current_time += aac->m_msec_per_frame;
aac->m_current_frame++;
} else {
aac->m_last_rtp_ts = ts;
aac->m_current_time = ts;
aac->m_current_frame = 0;
}
// Note - here m_current_time should pretty much always be >= rtpts.
// If we're not, we most likely want to stop and resync. We don't
// need to keep decoding - just decode this frame and indicate we
// need a resync... That should handle fast forwards... We need
// someway to handle reverses - perhaps if we're more than .5 seconds
// later...
}
if (aac->m_faad_inited == 0) {
/*
* If not initialized, do so.
*/
abort();
unsigned long freq;
unsigned char chans;
faacDecInit(aac->m_info,
(unsigned char *)buffer,
buflen,
&freq,
&chans);
aac->m_freq = freq;
aac->m_chans = chans;
aac->m_faad_inited = 1;
}
uint8_t *buff;
unsigned long samples;
bytes_consummed = buflen;
//aa_message(LOG_DEBUG, aaclib, "decoding %d bits", buflen * 8);
faacDecFrameInfo frame_info;
buff = (uint8_t *)faacDecDecode(aac->m_info,
&frame_info,
buffer,
buflen);
if (buff != NULL) {
bytes_consummed = frame_info.bytesconsumed;
#if 0
aa_message(LOG_DEBUG, aaclib, LLU" bytes %d samples %d",
ts, bytes_consummed, frame_info.samples);
#endif
if (aac->m_audio_inited != 0) {
int tempchans = frame_info.channels;
if (tempchans != aac->m_chans) {
aa_message(LOG_NOTICE, aaclib, "chupdate - chans from data is %d",
tempchans);
}
} else {
int tempchans = frame_info.channels;
if (tempchans == 0) {
aa_message(LOG_ERR, aaclib, "initializing aac, returned channels are 0");
aac->m_resync_with_header = 1;
aac->m_record_sync_time = 1;
return bytes_consummed;
}
aac->m_chans = tempchans;
aac->m_freq = frame_info.samplerate;
aac->m_vft->audio_configure(aac->m_ifptr,
aac->m_freq,
aac->m_chans,
AUDIO_S16SYS,
aac->m_output_frame_size);
uint8_t *now = aac->m_vft->audio_get_buffer(aac->m_ifptr);
aac->m_audio_inited = 1;
}
/*
* good result - give it to audio sync class
*/
#if DUMP_OUTPUT_TO_FILE
fwrite(buff, aac->m_output_frame_size * 4, 1, aac->m_outfile);
#endif
if (frame_info.samples != 0) {
aac->m_vft->audio_load_buffer(aac->m_ifptr,
buff,
frame_info.samples * 2,
aac->m_last_ts,
aac->m_resync_with_header);
if (aac->m_resync_with_header == 1) {
aac->m_resync_with_header = 0;
#ifdef DEBUG_SYNC
aa_message(LOG_DEBUG, aaclib, "Back to good at "LLU, aac->m_current_time);
#endif
}
}
} else {
aa_message(LOG_ERR, aaclib, "error return is %d", frame_info.error);
aac->m_resync_with_header = 1;
#ifdef DEBUG_SYNC
aa_message(LOG_ERR, aaclib, "Audio decode problem - at "LLU,
aac->m_current_time);
#endif
}
aac->m_last_ts = aac->m_current_time;
return (bytes_consummed);
}
/*****************************************************************************
* DecodeBlock:
*****************************************************************************/
static aout_buffer_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
block_t *p_block;
if( !pp_block || !*pp_block ) return NULL;
p_block = *pp_block;
if( p_block->i_flags&(BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
block_Release( p_block );
return NULL;
}
/* Remove ADTS header if we have decoder specific config */
if( p_dec->fmt_in.i_extra && p_block->i_buffer > 7 )
{
if( p_block->p_buffer[0] == 0xff &&
( p_block->p_buffer[1] & 0xf0 ) == 0xf0 ) /* syncword */
{ /* ADTS header present */
size_t i_header_size; /* 7 bytes (+ 2 bytes for crc) */
i_header_size = 7 + ( ( p_block->p_buffer[1] & 0x01 ) ? 0 : 2 );
/* FIXME: multiple blocks per frame */
if( p_block->i_buffer > i_header_size )
{
vlc_memcpy( p_block->p_buffer,
p_block->p_buffer + i_header_size,
p_block->i_buffer - i_header_size );
p_block->i_buffer -= i_header_size;
}
}
}
/* Append the block to the temporary buffer */
if( p_sys->i_buffer_size < p_sys->i_buffer + p_block->i_buffer )
{
p_sys->i_buffer_size = p_sys->i_buffer + p_block->i_buffer;
p_sys->p_buffer = realloc( p_sys->p_buffer, p_sys->i_buffer_size );
}
if( p_block->i_buffer > 0 )
{
vlc_memcpy( &p_sys->p_buffer[p_sys->i_buffer],
p_block->p_buffer, p_block->i_buffer );
p_sys->i_buffer += p_block->i_buffer;
p_block->i_buffer = 0;
}
if( p_dec->fmt_out.audio.i_rate == 0 && p_dec->fmt_in.i_extra > 0 )
{
/* We have a decoder config so init the handle */
unsigned long i_rate;
unsigned char i_channels;
if( faacDecInit2( p_sys->hfaad, p_dec->fmt_in.p_extra,
p_dec->fmt_in.i_extra,
&i_rate, &i_channels ) >= 0 )
{
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
aout_DateInit( &p_sys->date, i_rate );
}
}
if( p_dec->fmt_out.audio.i_rate == 0 && p_sys->i_buffer )
{
unsigned long i_rate;
unsigned char i_channels;
/* Init faad with the first frame */
if( faacDecInit( p_sys->hfaad,
p_sys->p_buffer, p_sys->i_buffer,
&i_rate, &i_channels ) < 0 )
{
block_Release( p_block );
return NULL;
}
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
aout_DateInit( &p_sys->date, i_rate );
}
if( p_block->i_pts != 0 && p_block->i_pts != aout_DateGet( &p_sys->date ) )
{
aout_DateSet( &p_sys->date, p_block->i_pts );
}
else if( !aout_DateGet( &p_sys->date ) )
{
/* We've just started the stream, wait for the first PTS. */
block_Release( p_block );
p_sys->i_buffer = 0;
return NULL;
}
/* Decode all data */
if( p_sys->i_buffer )
{
void *samples;
faacDecFrameInfo frame;
aout_buffer_t *p_out;
int i, j;
samples = faacDecDecode( p_sys->hfaad, &frame,
p_sys->p_buffer, p_sys->i_buffer );
if( frame.error > 0 )
{
msg_Warn( p_dec, "%s", faacDecGetErrorMessage( frame.error ) );
/* Flush the buffer */
p_sys->i_buffer = 0;
block_Release( p_block );
return NULL;
}
if( frame.channels <= 0 || frame.channels > 8 || frame.channels == 7 )
{
msg_Warn( p_dec, "invalid channels count: %i", frame.channels );
/* Flush the buffer */
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer,&p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
block_Release( p_block );
return NULL;
}
if( frame.samples <= 0 )
{
msg_Warn( p_dec, "decoded zero sample" );
/* Flush the buffer */
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer,&p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
block_Release( p_block );
return NULL;
}
/* We decoded a valid frame */
if( p_dec->fmt_out.audio.i_rate != frame.samplerate )
{
aout_DateInit( &p_sys->date, frame.samplerate );
aout_DateSet( &p_sys->date, p_block->i_pts );
}
p_block->i_pts = 0; /* PTS is valid only once */
p_dec->fmt_out.audio.i_rate = frame.samplerate;
p_dec->fmt_out.audio.i_channels = frame.channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[frame.channels];
/* Adjust stream info when dealing with SBR/PS */
if( p_sys->b_sbr != frame.sbr || p_sys->b_ps != frame.ps )
{
const char *psz_ext = (frame.sbr && frame.ps) ? "SBR+PS" :
frame.sbr ? "SBR" : "PS";
msg_Dbg( p_dec, "AAC %s (channels: %u, samplerate: %lu)",
psz_ext, frame.channels, frame.samplerate );
if( !p_dec->p_description )
p_dec->p_description = vlc_meta_New();
if( p_dec->p_description )
vlc_meta_AddExtra( p_dec->p_description, _("AAC extension"), psz_ext );
p_sys->b_sbr = frame.sbr; p_sys->b_ps = frame.ps;
}
/* Convert frame.channel_position to our own channel values */
p_dec->fmt_out.audio.i_physical_channels = 0;
for( i = 0; i < frame.channels; i++ )
{
/* Find the channel code */
for( j = 0; j < MAX_CHANNEL_POSITIONS; j++ )
{
if( frame.channel_position[i] == pi_channels_in[j] )
break;
}
if( j >= MAX_CHANNEL_POSITIONS )
{
msg_Warn( p_dec, "unknown channel ordering" );
/* Invent something */
j = i;
}
/* */
p_sys->pi_channel_positions[i] = pi_channels_out[j];
if( p_dec->fmt_out.audio.i_physical_channels & pi_channels_out[j] )
frame.channels--; /* We loose a duplicated channel */
else
p_dec->fmt_out.audio.i_physical_channels |= pi_channels_out[j];
}
p_dec->fmt_out.audio.i_original_channels =
p_dec->fmt_out.audio.i_physical_channels;
p_out = decoder_NewAudioBuffer(p_dec, frame.samples/frame.channels);
if( p_out == NULL )
{
p_sys->i_buffer = 0;
block_Release( p_block );
return NULL;
}
p_out->start_date = aout_DateGet( &p_sys->date );
p_out->end_date = aout_DateIncrement( &p_sys->date, frame.samples / frame.channels );
DoReordering( (uint32_t *)p_out->p_buffer, samples,
frame.samples / frame.channels, frame.channels,
p_sys->pi_channel_positions );
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer, &p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
return p_out;
}
block_Release( p_block );
return NULL;
}
int decodeAACfile(char *sndfile, int def_srate, aac_dec_opt *opt)
{
int tagsize;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
FILE *infile;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
int first_time = 1;
/* declare variables for buffering */
DEC_BUFF_VARS
infile = fopen(opt->filename, "rb");
if (infile == NULL)
{
/* unable to open file */
error_handler("Error opening file: %s\n", opt->filename);
return 1;
}
INIT_BUFF(infile)
tagsize = id3v2_tag(buffer);
if (tagsize)
{
UPDATE_BUFF_SKIP(tagsize)
}
hDecoder = faacDecOpen();
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
config = faacDecGetCurrentConfiguration(hDecoder);
if (def_srate)
config->defSampleRate = def_srate;
config->defObjectType = opt->object_type;
config->outputFormat = opt->output_format;
faacDecSetConfiguration(hDecoder, config);
if ((bytesconsumed = faacDecInit(hDecoder, buffer, bytes_in_buffer,
&samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
error_handler("Error initializing decoder library.\n");
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 1;
}
buffer_index += bytesconsumed;
do
{
/* update buffer */
UPDATE_BUFF_READ
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer, bytes_in_buffer);
/* update buffer indices */
UPDATE_BUFF_IDX(frameInfo)
if (frameInfo.error > 0)
{
error_handler("Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
opt->progress_update((long)fileread, buffer_index);
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(opt->decode_mode == 0)
{
if (Set_WIN_Params (INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
frameInfo.channels) < 0)
{
error_handler("\nCan't access %s\n", "WAVE OUT");
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return (0);
}
}
else
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
opt->output_format, opt->file_type, aacChannelConfig2wavexChannelMask(&frameInfo));
if (aufile == NULL)
{
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 0;
}
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
if(opt->decode_mode == 0)
WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
else
write_audio_file(aufile, sample_buffer, frameInfo.samples, 0);
}
if (buffer_index >= fileread)
sample_buffer = NULL; /* to make sure it stops now */
if(stop_decoding)
break;
} while (sample_buffer != NULL);
faacDecClose(hDecoder);
fclose(infile);
if(opt->decode_mode == 0)
WIN_Audio_close();
else
{
if (!first_time)
close_audio_file(aufile);
}
END_BUFF
return frameInfo.error;
}
int AacPcm::processData(MediaInfo *infos, ChunkList *chunk_list, bool *killswitch)
{
DWORD read,
tmp,
shorts_decoded=0;
long result=0;
ChunkInfosI *ci = new ChunkInfosI();
ci->addInfo("srate", dwSamprate);
ci->addInfo("bps", 16);
ci->addInfo("nch", dwChannels);
if(bytes_into_buffer<0)
ERROR_processData(0);
svc_fileReader *reader = infos->getReader();
if(!reader)
ERROR_processData("File doesn\'t exists");
do
{
if(bytes_consumed>0 && bytes_into_buffer>=0)
{
if(bytes_into_buffer)
memcpy(buffer,buffer+bytes_consumed,bytes_into_buffer);
if(bytes_read<lSize)
{
if(bytes_read+bytes_consumed<lSize)
tmp=bytes_consumed;
else
tmp=lSize-bytes_read;
read=reader->read((char *)buffer+bytes_into_buffer, tmp);
if(read==tmp)
{
bytes_read+=read;
bytes_into_buffer+=read;
}
else
infos->status("Read failed!"); // continue until bytes_into_buffer<1
}
else
if(bytes_into_buffer)
memset(buffer+bytes_into_buffer, 0, bytes_consumed);
}
if(bytes_into_buffer<1)
if(bytes_read<lSize)
ERROR_processData("Buffer empty!")
else
return 0;
result=faacDecDecode(hDecoder, buffer, &bytes_consumed, (short*)bufout, &shorts_decoded);
bytes_into_buffer-=bytes_consumed;
}while(!shorts_decoded || result==FAAD_OK_CHUPDATE);
if(result==FAAD_FATAL_ERROR || result==FAAD_ERROR)
ERROR_processData("FAAD_FATAL_ERROR or FAAD_ERROR");
shorts_decoded*=sizeof(short);
chunk_list->setChunk("PCM", bufout, shorts_decoded, ci);
// fwrite(bufout,1,bytesDec,fil);
if(!shorts_decoded)
ERROR_processData(0);
return 1;
}
/*
* Decode task call for FAAC
*/
static int aac_decode (codec_data_t *ptr,
frame_timestamp_t *ts,
int from_rtp,
int *sync_frame,
uint8_t *buffer,
uint32_t buflen,
void *userdata)
{
aac_codec_t *aac = (aac_codec_t *)ptr;
unsigned long bytes_consummed;
int bits = -1;
// struct timezone tz;
uint32_t freq_timestamp;
freq_timestamp = ts->audio_freq_timestamp;
if (ts->audio_freq != aac->m_freq) {
freq_timestamp = convert_timescale(freq_timestamp,
ts->audio_freq,
aac->m_freq);
}
if (aac->m_record_sync_time) {
aac->m_current_frame = 0;
aac->m_record_sync_time = 0;
aac->m_current_time = ts->msec_timestamp;
aac->m_last_rtp_ts = ts->msec_timestamp;
} else {
if (aac->m_last_rtp_ts == ts->msec_timestamp) {
aac->m_current_frame++;
aac->m_current_time = aac->m_last_rtp_ts;
aac->m_current_time +=
aac->m_output_frame_size * aac->m_current_frame *
TO_U64(1000) / aac->m_freq;
freq_timestamp += aac->m_output_frame_size * aac->m_current_frame;
} else {
aac->m_last_rtp_ts = ts->msec_timestamp;
aac->m_current_time = ts->msec_timestamp;
aac->m_current_frame = 0;
}
// Note - here m_current_time should pretty much always be >= rtpts.
// If we're not, we most likely want to stop and resync. We don't
// need to keep decoding - just decode this frame and indicate we
// need a resync... That should handle fast forwards... We need
// someway to handle reverses - perhaps if we're more than .5 seconds
// later...
}
if (aac->m_faad_inited == 0) {
/*
* If not initialized, do so.
*/
unsigned long freq, chans;
faacDecInit(aac->m_info,
(unsigned char *)buffer,
&freq,
&chans);
aac->m_freq = freq;
aac->m_chans = chans;
aac->m_faad_inited = 1;
}
uint8_t *buff;
/*
* Get an audio buffer
*/
if (aac->m_audio_inited == 0) {
buff = aac->m_temp_buff;
} else {
buff = aac->m_vft->audio_get_buffer(aac->m_ifptr,
freq_timestamp,
aac->m_current_time);
}
if (buff == NULL) {
//player_debug_message("Can't get buffer in aa");
return (0);
}
unsigned long samples;
bytes_consummed = buflen;
//aa_message(LOG_DEBUG, aaclib, "decoding %d bits", buflen * 8);
bits = faacDecDecode(aac->m_info,
(unsigned char *)buffer,
&bytes_consummed,
(short *)buff,
&samples);
switch (bits) {
case FAAD_OK_CHUPDATE:
if (aac->m_audio_inited != 0) {
int tempchans = faacDecGetProgConfig(aac->m_info, NULL);
if (tempchans != aac->m_chans) {
aa_message(LOG_NOTICE, aaclib, "chupdate - chans from data is %d",
tempchans);
}
}
// fall through...
case FAAD_OK:
if (aac->m_audio_inited == 0) {
int tempchans = faacDecGetProgConfig(aac->m_info, NULL);
if (tempchans == 0) {
aac->m_resync_with_header = 1;
aac->m_record_sync_time = 1;
return bytes_consummed;
}
if (tempchans != aac->m_chans) {
aa_message(LOG_NOTICE, aaclib, "chans from data is %d conf %d",
tempchans, aac->m_chans);
aac->m_chans = tempchans;
}
aac->m_vft->audio_configure(aac->m_ifptr,
aac->m_freq,
aac->m_chans,
AUDIO_FMT_S16,
aac->m_output_frame_size);
uint8_t *now = aac->m_vft->audio_get_buffer(aac->m_ifptr,
freq_timestamp,
aac->m_current_time);
if (now != NULL) {
memcpy(now, buff, tempchans * aac->m_output_frame_size * sizeof(int16_t));
}
aac->m_audio_inited = 1;
}
/*
* good result - give it to audio sync class
*/
#if DUMP_OUTPUT_TO_FILE
fwrite(buff, aac->m_output_frame_size * 4, 1, aac->m_outfile);
#endif
aac->m_vft->audio_filled_buffer(aac->m_ifptr);
if (aac->m_resync_with_header == 1) {
aac->m_resync_with_header = 0;
#ifdef DEBUG_SYNC
aa_message(LOG_DEBUG, aaclib, "Back to good at "U64, aac->m_current_time);
#endif
}
break;
default:
aa_message(LOG_ERR, aaclib, "Bits return is %d", bits);
aac->m_resync_with_header = 1;
#ifdef DEBUG_SYNC
aa_message(LOG_ERR, aaclib, "Audio decode problem - at "U64,
aac->m_current_time);
#endif
break;
}
return (bytes_consummed);
}
static void *mp4Decode(void *args)
{
MP4FileHandle mp4file;
pthread_mutex_lock(&mutex);
seekPosition = -1;
bPlaying = TRUE;
if(!(mp4file = MP4Read(args, 0))){
mp4cfg.file_type = FILE_AAC;
MP4Close(mp4file);
}else{
mp4cfg.file_type = FILE_MP4;
}
if(mp4cfg.file_type == FILE_MP4){
// We are reading a MP4 file
gint mp4track;
if((mp4track = getAACTrack(mp4file)) < 0){
//TODO: check here for others Audio format.....
g_print("Unsupported Audio track type\n");
g_free(args);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}else{
faacDecHandle decoder;
unsigned char *buffer = NULL;
guint bufferSize = 0;
gulong samplerate;
guchar channels;
guint avgBitrate;
MP4Duration duration;
gulong msDuration;
MP4SampleId numSamples;
MP4SampleId sampleID = 1;
decoder = faacDecOpen();
MP4GetTrackESConfiguration(mp4file, mp4track, &buffer, &bufferSize);
if(!buffer){
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if(faacDecInit2(decoder, buffer, bufferSize, &samplerate, &channels)<0){
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
g_free(buffer);
if(channels == 0){
g_print("Number of Channels not supported\n");
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
duration = MP4GetTrackDuration(mp4file, mp4track);
msDuration = MP4ConvertFromTrackDuration(mp4file, mp4track, duration,
MP4_MSECS_TIME_SCALE);
numSamples = MP4GetTrackNumberOfSamples(mp4file, mp4track);
mp4_ip.output->open_audio(FMT_S16_NE, samplerate, channels);
mp4_ip.output->flush(0);
mp4_ip.set_info(args, msDuration, -1, samplerate/1000, channels);
g_print("MP4 - %d channels @ %d Hz\n", channels, samplerate);
while(bPlaying){
void* sampleBuffer;
faacDecFrameInfo frameInfo;
gint rc;
if(seekPosition!=-1){
duration = MP4ConvertToTrackDuration(mp4file,
mp4track,
seekPosition*1000,
MP4_MSECS_TIME_SCALE);
sampleID = MP4GetSampleIdFromTime(mp4file, mp4track, duration, 0);
mp4_ip.output->flush(seekPosition*1000);
seekPosition = -1;
}
buffer=NULL;
bufferSize=0;
if(sampleID > numSamples){
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
rc = MP4ReadSample(mp4file, mp4track, sampleID++, &buffer, &bufferSize,
NULL, NULL, NULL, NULL);
//g_print("%d/%d\n", sampleID-1, numSamples);
if((rc==0) || (buffer== NULL)){
g_print("MP4: read error\n");
sampleBuffer = NULL;
sampleID=0;
mp4_ip.output->buffer_free();
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}else{
sampleBuffer = faacDecDecode(decoder, &frameInfo, buffer, bufferSize);
if(frameInfo.error > 0){
g_print("MP4: %s\n",
faacDecGetErrorMessage(frameInfo.error));
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if(buffer){
g_free(buffer); buffer=NULL; bufferSize=0;
}
while(bPlaying && mp4_ip.output->buffer_free()<frameInfo.samples<<1)
xmms_usleep(30000);
}
mp4_ip.add_vis_pcm(mp4_ip.output->written_time(),
FMT_S16_NE,
channels,
frameInfo.samples<<1,
sampleBuffer);
mp4_ip.output->write_audio(sampleBuffer, frameInfo.samples<<1);
}
while(bPlaying && mp4_ip.output->buffer_free()){
xmms_usleep(10000);
}
mp4_ip.output->close_audio();
g_free(args);
faacDecClose(decoder);
MP4Close(mp4file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
} else{
// WE ARE READING AN AAC FILE
FILE *file = NULL;
faacDecHandle decoder = 0;
guchar *buffer = 0;
gulong bufferconsumed = 0;
gulong samplerate = 0;
guchar channels;
gulong buffervalid = 0;
TitleInput* input;
gchar *temp = g_strdup(args);
gchar *ext = strrchr(temp, '.');
gchar *xmmstitle = NULL;
faacDecConfigurationPtr config;
if((file = fopen(args, "rb")) == 0){
g_print("AAC: can't find file %s\n", args);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if((decoder = faacDecOpen()) == NULL){
g_print("AAC: Open Decoder Error\n");
fclose(file);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
config = faacDecGetCurrentConfiguration(decoder);
config->useOldADTSFormat = 0;
faacDecSetConfiguration(decoder, config);
if((buffer = g_malloc(BUFFER_SIZE)) == NULL){
g_print("AAC: error g_malloc\n");
fclose(file);
bPlaying = FALSE;
faacDecClose(decoder);
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
if((buffervalid = fread(buffer, 1, BUFFER_SIZE, file))==0){
g_print("AAC: Error reading file\n");
g_free(buffer);
fclose(file);
bPlaying = FALSE;
faacDecClose(decoder);
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
XMMS_NEW_TITLEINPUT(input);
input->file_name = g_basename(temp);
input->file_ext = ext ? ext+1 : NULL;
input->file_path = temp;
if(!strncmp(buffer, "ID3", 3)){
gint size = 0;
fseek(file, 0, SEEK_SET);
size = (buffer[6]<<21) | (buffer[7]<<14) | (buffer[8]<<7) | buffer[9];
size+=10;
fread(buffer, 1, size, file);
buffervalid = fread(buffer, 1, BUFFER_SIZE, file);
}
xmmstitle = xmms_get_titlestring(xmms_get_gentitle_format(), input);
if(xmmstitle == NULL)
xmmstitle = g_strdup(input->file_name);
if(temp) g_free(temp);
if(input->performer) g_free(input->performer);
if(input->album_name) g_free(input->album_name);
if(input->track_name) g_free(input->track_name);
if(input->genre) g_free(input->genre);
g_free(input);
bufferconsumed = faacDecInit(decoder,
buffer,
buffervalid,
&samplerate,
&channels);
if(mp4_ip.output->open_audio(FMT_S16_NE,samplerate,channels) == FALSE){
g_print("AAC: Output Error\n");
g_free(buffer); buffer=0;
faacDecClose(decoder);
fclose(file);
mp4_ip.output->close_audio();
/*
if(positionTable){
g_free(positionTable); positionTable=0;
}
*/
g_free(xmmstitle);
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
//if(bSeek){
//mp4_ip.set_info(xmmstitle, lenght*1000, -1, samplerate, channels);
//}else{
mp4_ip.set_info(xmmstitle, -1, -1, samplerate, channels);
//}
mp4_ip.output->flush(0);
while(bPlaying && buffervalid > 0){
faacDecFrameInfo finfo;
unsigned long samplesdecoded;
char* sample_buffer = NULL;
/*
if(bSeek && seekPosition!=-1){
fseek(file, positionTable[seekPosition], SEEK_SET);
bufferconsumed=0;
buffervalid = fread(buffer, 1, BUFFER_SIZE, file);
aac_ip.output->flush(seekPosition*1000);
seekPosition=-1;
}
*/
if(bufferconsumed > 0){
memmove(buffer, &buffer[bufferconsumed], buffervalid-bufferconsumed);
buffervalid -= bufferconsumed;
buffervalid += fread(&buffer[buffervalid], 1,
BUFFER_SIZE-buffervalid, file);
bufferconsumed = 0;
}
sample_buffer = faacDecDecode(decoder, &finfo, buffer, buffervalid);
if(finfo.error){
config = faacDecGetCurrentConfiguration(decoder);
if(config->useOldADTSFormat != 1){
faacDecClose(decoder);
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->useOldADTSFormat = 1;
faacDecSetConfiguration(decoder, config);
finfo.bytesconsumed=0;
finfo.samples = 0;
faacDecInit(decoder,
buffer,
buffervalid,
&samplerate,
&channels);
}else{
g_print("FAAD2 Warning %s\n", faacDecGetErrorMessage(finfo.error));
buffervalid = 0;
}
}
bufferconsumed += finfo.bytesconsumed;
samplesdecoded = finfo.samples;
if((samplesdecoded<=0) && !sample_buffer){
g_print("AAC: error sample decoding\n");
continue;
}
while(bPlaying && mp4_ip.output->buffer_free() < (samplesdecoded<<1)){
xmms_usleep(10000);
}
mp4_ip.add_vis_pcm(mp4_ip.output->written_time(),
FMT_S16_LE, channels,
samplesdecoded<<1, sample_buffer);
mp4_ip.output->write_audio(sample_buffer, samplesdecoded<<1);
}
while(bPlaying && mp4_ip.output->buffer_playing()){
xmms_usleep(10000);
}
mp4_ip.output->buffer_free();
mp4_ip.output->close_audio();
bPlaying = FALSE;
g_free(buffer);
faacDecClose(decoder);
g_free(xmmstitle);
fclose(file);
seekPosition = -1;
/*
if(positionTable){
g_free(positionTable); positionTable=0;
}
*/
bPlaying = FALSE;
pthread_mutex_unlock(&mutex);
pthread_exit(NULL);
}
}
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
int j = 0, len = 0, last_dec_len = 1, errors = 0;
void *faac_sample_buffer;
while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) {
/* update buffer for raw aac streams: */
if(!sh->codecdata_len)
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
#ifdef DUMP_AAC_COMPRESSED
{int i;
for (i = 0; i < 16; i++)
printf ("%02X ", sh->a_in_buffer[i]);
printf ("\n");}
#endif
if(!sh->codecdata_len){
// raw aac stream:
do {
faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len);
/* update buffer index after faacDecDecode */
if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) {
sh->a_in_buffer_len=0;
} else {
sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
}
if(faac_finfo.error > 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n",
faacDecGetErrorMessage(faac_finfo.error));
if (sh->a_in_buffer_len <= 0) {
errors = MAX_FAAD_ERRORS;
break;
}
sh->a_in_buffer_len--;
memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
aac_sync(sh);
errors++;
} else
break;
} while(errors < MAX_FAAD_ERRORS);
} else {
// packetized (.mp4) aac stream:
unsigned char* bufptr=NULL;
double pts;
int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
if(buflen<=0) break;
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
}
//for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
if(faac_finfo.error > 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
faacDecGetErrorMessage(faac_finfo.error));
} else if (faac_finfo.samples == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
} else {
/* XXX: samples already multiplied by channels! */
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n",
sh->samplesize*faac_finfo.samples);
memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
last_dec_len = sh->samplesize*faac_finfo.samples;
len += last_dec_len;
sh->pts_bytes += last_dec_len;
//printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed);
}
}
return len;
}
/*****************************************************************************
* DecodeBlock:
*****************************************************************************/
static block_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
block_t *p_block;
if( !pp_block || !*pp_block ) return NULL;
p_block = *pp_block;
if( p_block->i_flags&(BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
block_Release( p_block );
return NULL;
}
/* Remove ADTS header if we have decoder specific config */
if( p_dec->fmt_in.i_extra && p_block->i_buffer > 7 )
{
if( p_block->p_buffer[0] == 0xff &&
( p_block->p_buffer[1] & 0xf0 ) == 0xf0 ) /* syncword */
{ /* ADTS header present */
size_t i_header_size; /* 7 bytes (+ 2 bytes for crc) */
i_header_size = 7 + ( ( p_block->p_buffer[1] & 0x01 ) ? 0 : 2 );
/* FIXME: multiple blocks per frame */
if( p_block->i_buffer > i_header_size )
{
p_block->p_buffer += i_header_size;
p_block->i_buffer -= i_header_size;
}
}
}
/* Append the block to the temporary buffer */
if( p_sys->i_buffer_size < p_sys->i_buffer + p_block->i_buffer )
{
size_t i_buffer_size = p_sys->i_buffer + p_block->i_buffer;
uint8_t *p_buffer = realloc( p_sys->p_buffer, i_buffer_size );
if( p_buffer )
{
p_sys->i_buffer_size = i_buffer_size;
p_sys->p_buffer = p_buffer;
}
else
{
p_block->i_buffer = 0;
}
}
if( p_block->i_buffer > 0 )
{
memcpy( &p_sys->p_buffer[p_sys->i_buffer],
p_block->p_buffer, p_block->i_buffer );
p_sys->i_buffer += p_block->i_buffer;
p_block->i_buffer = 0;
}
if( p_dec->fmt_out.audio.i_rate == 0 && p_dec->fmt_in.i_extra > 0 )
{
/* We have a decoder config so init the handle */
unsigned long i_rate;
unsigned char i_channels;
if( faacDecInit2( p_sys->hfaad, p_dec->fmt_in.p_extra,
p_dec->fmt_in.i_extra,
&i_rate, &i_channels ) >= 0 )
{
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
}
if( p_dec->fmt_out.audio.i_rate == 0 && p_sys->i_buffer )
{
unsigned long i_rate;
unsigned char i_channels;
/* Init faad with the first frame */
if( faacDecInit( p_sys->hfaad,
p_sys->p_buffer, p_sys->i_buffer,
&i_rate, &i_channels ) < 0 )
{
block_Release( p_block );
return NULL;
}
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
if( p_block->i_pts > VLC_TS_INVALID && p_block->i_pts != date_Get( &p_sys->date ) )
{
date_Set( &p_sys->date, p_block->i_pts );
}
else if( !date_Get( &p_sys->date ) )
{
/* We've just started the stream, wait for the first PTS. */
block_Release( p_block );
p_sys->i_buffer = 0;
return NULL;
}
/* Decode all data */
if( p_sys->i_buffer )
{
void *samples;
faacDecFrameInfo frame;
block_t *p_out;
samples = faacDecDecode( p_sys->hfaad, &frame,
p_sys->p_buffer, p_sys->i_buffer );
if( frame.error > 0 )
{
msg_Warn( p_dec, "%s", faacDecGetErrorMessage( frame.error ) );
if( frame.error == 21 || frame.error == 12 )
{
/*
* Once an "Unexpected channel configuration change"
* or a "Invalid number of channels" error
* occurs, it will occurs afterwards, and we got no sound.
* Reinitialization of the decoder is required.
*/
unsigned long i_rate;
unsigned char i_channels;
faacDecHandle *hfaad;
faacDecConfiguration *cfg,*oldcfg;
oldcfg = faacDecGetCurrentConfiguration( p_sys->hfaad );
hfaad = faacDecOpen();
cfg = faacDecGetCurrentConfiguration( hfaad );
if( oldcfg->defSampleRate )
cfg->defSampleRate = oldcfg->defSampleRate;
cfg->defObjectType = oldcfg->defObjectType;
cfg->outputFormat = oldcfg->outputFormat;
faacDecSetConfiguration( hfaad, cfg );
if( faacDecInit( hfaad, p_sys->p_buffer, p_sys->i_buffer,
&i_rate,&i_channels ) < 0 )
{
/* reinitialization failed */
faacDecClose( hfaad );
faacDecSetConfiguration( p_sys->hfaad, oldcfg );
}
else
{
faacDecClose( p_sys->hfaad );
p_sys->hfaad = hfaad;
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[i_channels];
date_Init( &p_sys->date, i_rate, 1 );
}
}
/* Flush the buffer */
p_sys->i_buffer = 0;
block_Release( p_block );
return NULL;
}
if( frame.channels <= 0 || frame.channels > 8 || frame.channels == 7 )
{
msg_Warn( p_dec, "invalid channels count: %i", frame.channels );
/* Flush the buffer */
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer,&p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
block_Release( p_block );
return NULL;
}
if( frame.samples <= 0 )
{
msg_Warn( p_dec, "decoded zero sample" );
/* Flush the buffer */
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer,&p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
block_Release( p_block );
return NULL;
}
/* We decoded a valid frame */
if( p_dec->fmt_out.audio.i_rate != frame.samplerate )
{
date_Init( &p_sys->date, frame.samplerate, 1 );
date_Set( &p_sys->date, p_block->i_pts );
}
p_block->i_pts = VLC_TS_INVALID; /* PTS is valid only once */
p_dec->fmt_out.audio.i_rate = frame.samplerate;
p_dec->fmt_out.audio.i_channels = frame.channels;
/* Adjust stream info when dealing with SBR/PS */
bool b_sbr = (frame.sbr == 1) || (frame.sbr == 2);
if( p_sys->b_sbr != b_sbr || p_sys->b_ps != frame.ps )
{
const char *psz_ext = (b_sbr && frame.ps) ? "SBR+PS" :
b_sbr ? "SBR" : "PS";
msg_Dbg( p_dec, "AAC %s (channels: %u, samplerate: %lu)",
psz_ext, frame.channels, frame.samplerate );
if( !p_dec->p_description )
p_dec->p_description = vlc_meta_New();
if( p_dec->p_description )
vlc_meta_AddExtra( p_dec->p_description, _("AAC extension"), psz_ext );
p_sys->b_sbr = b_sbr;
p_sys->b_ps = frame.ps;
}
/* Convert frame.channel_position to our own channel values */
p_dec->fmt_out.audio.i_physical_channels = 0;
const uint32_t nbChannels = frame.channels;
unsigned j;
for( unsigned i = 0; i < nbChannels; i++ )
{
/* Find the channel code */
for( j = 0; j < MAX_CHANNEL_POSITIONS; j++ )
{
if( frame.channel_position[i] == pi_channels_in[j] )
break;
}
if( j >= MAX_CHANNEL_POSITIONS )
{
msg_Warn( p_dec, "unknown channel ordering" );
/* Invent something */
j = i;
}
/* */
p_sys->pi_channel_positions[i] = pi_channels_out[j];
if( p_dec->fmt_out.audio.i_physical_channels & pi_channels_out[j] )
frame.channels--; /* We loose a duplicated channel */
else
p_dec->fmt_out.audio.i_physical_channels |= pi_channels_out[j];
}
if ( nbChannels != frame.channels )
{
p_dec->fmt_out.audio.i_physical_channels
= p_dec->fmt_out.audio.i_original_channels
= pi_channels_guessed[nbChannels];
}
else
{
p_dec->fmt_out.audio.i_original_channels =
p_dec->fmt_out.audio.i_physical_channels;
}
p_dec->fmt_out.audio.i_channels = nbChannels;
p_out = decoder_NewAudioBuffer( p_dec, frame.samples / nbChannels );
if( p_out == NULL )
{
p_sys->i_buffer = 0;
block_Release( p_block );
return NULL;
}
p_out->i_pts = date_Get( &p_sys->date );
p_out->i_length = date_Increment( &p_sys->date,
frame.samples / nbChannels )
- p_out->i_pts;
DoReordering( (uint32_t *)p_out->p_buffer, samples,
frame.samples / nbChannels, nbChannels,
p_sys->pi_channel_positions );
p_sys->i_buffer -= frame.bytesconsumed;
if( p_sys->i_buffer > 0 )
{
memmove( p_sys->p_buffer, &p_sys->p_buffer[frame.bytesconsumed],
p_sys->i_buffer );
}
return p_out;
}
block_Release( p_block );
return NULL;
}
int decodeMP4file(char *sndfile, aac_dec_opt *opt)
{
int track;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
MP4FileHandle infile;
MP4SampleId sampleId, numSamples;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
unsigned char *buffer;
int buffer_size;
int first_time = 1;
hDecoder = faacDecOpen();
infile = MP4Read(opt->filename, 0);
if (!infile)
{
/* unable to open file */
error_handler("Error opening file: %s\n", opt->filename);
return 1;
}
if ((track = GetAACTrack(infile)) < 0)
{
error_handler("Unable to find correct AAC sound track in the MP4 file.\n");
MP4Close(infile);
return 1;
}
buffer = NULL;
buffer_size = 0;
MP4GetTrackESConfiguration(infile, track, &buffer, &buffer_size);
if(faacDecInit2(hDecoder, buffer, buffer_size, &samplerate, &channels) < 0)
{
/* If some error initializing occured, skip the file */
error_handler("Error initializing decoder library.\n");
faacDecClose(hDecoder);
MP4Close(infile);
return 1;
}
if (buffer)
free(buffer);
numSamples = MP4GetTrackNumberOfSamples(infile, track);
for (sampleId = 1; sampleId <= numSamples; sampleId++)
{
int rc;
/* get access unit from MP4 file */
buffer = NULL;
buffer_size = 0;
rc = MP4ReadSample(infile, track, sampleId, &buffer, &buffer_size, NULL, NULL, NULL, NULL);
if (rc == 0)
{
error_handler("Reading from MP4 file failed.\n");
faacDecClose(hDecoder);
MP4Close(infile);
return 1;
}
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer, buffer_size);
if (buffer)
free(buffer);
opt->progress_update((long)numSamples, sampleId);
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(opt->decode_mode == 0)
{
if (Set_WIN_Params (INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
frameInfo.channels) < 0)
{
error_handler("\nCan't access %s\n", "WAVE OUT");
faacDecClose(hDecoder);
MP4Close(infile);
return (0);
}
}
else
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
opt->output_format, opt->file_type, aacChannelConfig2wavexChannelMask(&frameInfo));
if (aufile == NULL)
{
faacDecClose(hDecoder);
MP4Close(infile);
return 0;
}
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
if(opt->decode_mode == 0)
WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
else
write_audio_file(aufile, sample_buffer, frameInfo.samples, 0);
}
if (frameInfo.error > 0)
{
error_handler("Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
break;
}
if(stop_decoding)
break;
}
faacDecClose(hDecoder);
MP4Close(infile);
if(opt->decode_mode == 0)
WIN_Audio_close();
else
{
if (!first_time)
close_audio_file(aufile);
}
return frameInfo.error;
}
uint8_t ADM_faad::run( uint8_t * ptr, uint32_t nbIn,
uint8_t * outptr, uint32_t * nbOut,ADM_ChannelMatrix *matrix)
{
long int res;
void *outbuf;
faacDecFrameInfo info;
int max=0;
#ifdef OLD_FAAD_PROTO
unsigned long int srate;
#else
uint32_t srate;
#endif
unsigned char chan=0;
uint8_t first=0;
ADM_assert(_instance);
*nbOut=0;
if(!_inited) // we still have'nt initialized the codec
{
// Try
printf("Trying with %d bytes\n",nbIn);
res=faacDecInit(_instance,ptr,nbIn,&srate,&chan);
if(res>=0)
{
printf("Faad Inited : rate:%d chan:%d off:%ld\n",srate,chan,res);
_inited=1;
first=1;
_inbuffer=0;
ptr+=res;
nbIn-=res;
}
}
if(!_inited)
{
printf("No dice...\n");
return 1;
}
// The codec is initialized, feed him
do
{
max=FAAD_BUFFER-_inbuffer;
if(nbIn<max) max=nbIn;
memcpy(_buffer+_inbuffer,ptr,max);
ptr+=max;
nbIn-=max;
_inbuffer+=max;
/*
if(_inbuffer<FAAD_MIN_STREAMSIZE*2)
{
return 1;
}*/
memset(&info,0,sizeof(info));
//printf("%x %x\n",_buffer[0],_buffer[1]);
outbuf = faacDecDecode(_instance, &info, _buffer, _inbuffer);
if(info.error)
{
printf("Faad: Error %d :%s\n",info.error,
faacDecGetErrorMessage(info.error));
/*
xin=info.bytesconsumed ;
if(!xin) xin=1;
memmove(_buffer,_buffer+xin,_inbuffer-xin);
_inbuffer-=xin;
return 1;*/
return 0;
}
if(first)
{
printf("Channels : %d\n",info.channels);
printf("Frequency: %d\n",info.samplerate);
printf("SBR : %d\n",info.sbr);
}
xin=info.bytesconsumed ;
if(xin>_inbuffer) xin=0;
xout=info.samples*2;
//printf("in:%d out:%d\n",xin,xout);
memmove(_buffer,_buffer+xin,_inbuffer-xin);
_inbuffer-=xin;
if(xout)
{
memcpy(outptr,outbuf,xout);
*nbOut+=xout;
outptr+=xout;
}
}while(nbIn);
return 1;
}