/*------------------------------------------------------------------------------
* Initialize the encoder
*----------------------------------------------------------------------------*/
void
VorbisLibEncoder :: init ( CastSink * sink,
unsigned int outMaxBitrate )
throw ( Exception )
{
this->sink = sink;
this->outMaxBitrate = outMaxBitrate;
if ( getInBitsPerSample() != 16 && getInBitsPerSample() != 8 ) {
throw Exception( __FILE__, __LINE__,
"specified bits per sample not supported",
getInBitsPerSample() );
}
if ( getInChannel() != 1 && getInChannel() != 2 ) {
throw Exception( __FILE__, __LINE__,
"unsupported number of channels for the encoder",
getInChannel() );
}
if ( getOutSampleRate() == getInSampleRate() ) {
resampleRatio = 1;
converter = 0;
} else {
resampleRatio = ( (double) getOutSampleRate() /
(double) getInSampleRate() );
// Determine if we can use linear interpolation.
// The inverse of the ratio must be a power of two for linear mode to
// be of sufficient quality.
bool useLinear = true;
double inverse = 1 / resampleRatio;
int integer = (int) inverse;
// Check that the inverse of the ratio is an integer
if( integer == inverse ) {
while( useLinear && integer ) { // Loop through the bits
// If the lowest order bit is not the only one set
if( integer & 1 && integer != 1 ) {
// Not a power of two; cannot use linear
useLinear = false;
} else {
// Shift all the bits over and try again
integer >>= 1;
}
}
} else {
useLinear = false;
}
// If we get here and useLinear is still true, then we have
// a power of two.
// open the aflibConverter in
// - high quality
// - linear or quadratic (non-linear) based on algorithm
// - not filter interpolation
converter = new aflibConverter( true, useLinear, false);
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
VorbisLibEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() ) {
return 0;
}
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int channels = getInChannel();
if ( channels != 1 && channels != 2 ) {
throw Exception( __FILE__, __LINE__,
"unsupported number of channels for the encoder",
channels );
}
unsigned int sampleSize = (bitsPerSample / 8) * channels;
unsigned char * b = (unsigned char*) buf;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
float ** vorbisBuffer;
// convert the byte-based raw input into a short buffer
// with channels still interleaved
unsigned int totalSamples = nSamples * channels;
short int shortBuffer[totalSamples];
Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
if ( converter ) {
// resample if needed
int inCount = totalSamples;
int outCount = (int) (inCount * resampleRatio);
short int resampledBuffer[outCount * channels];
int converted;
converted = converter->resample( inCount,
outCount,
shortBuffer,
resampledBuffer );
vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState,
converted / channels);
Util::conv( resampledBuffer, converted, vorbisBuffer, channels);
vorbis_analysis_wrote( &vorbisDspState, converted / channels);
} else {
vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState, nSamples);
Util::conv( shortBuffer, totalSamples, vorbisBuffer, channels);
vorbis_analysis_wrote( &vorbisDspState, nSamples);
}
vorbisBlocksOut();
return processed;
}
/*------------------------------------------------------------------------------
* Open an encoding session
*----------------------------------------------------------------------------*/
bool
aacPlusEncoder :: open ( void )
throw ( Exception )
{
if ( isOpen() ) {
close();
}
// open the underlying sink
if ( !sink->open() ) {
throw Exception( __FILE__, __LINE__,
"aacplus lib opening underlying sink error");
}
reportEvent(1, "Using aacplus codec version", "720 3gpp");
bitrate = getOutBitrate() * 1000;
bandwidth = 0;
useParametricStereo = 0;
numAncDataBytes=0;
coreWriteOffset = 0;
envReadOffset = 0;
writeOffset = INPUT_DELAY*MAX_CHANNELS;
writtenSamples = 0;
aacEnc = NULL;
hEnvEnc=NULL;
/* set up basic parameters for aacPlus codec */
AacInitDefaultConfig(&config);
nChannelsAAC = nChannelsSBR = getOutChannel();
if ( (getInChannel() == 2) && (bitrate >= 16000) && (bitrate < 44001) ) {
useParametricStereo = 1;
nChannelsAAC = 1;
nChannelsSBR = 2;
reportEvent(10, "use Parametric Stereo");
envReadOffset = (MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS;
coreWriteOffset = CORE_INPUT_OFFSET_PS;
writeOffset = envReadOffset;
} else {
/* set up 2:1 downsampling */
InitIIR21_Resampler(&(IIR21_reSampler[0]));
InitIIR21_Resampler(&(IIR21_reSampler[1]));
if(IIR21_reSampler[0].delay > MAX_DS_FILTER_DELAY)
throw Exception(__FILE__, __LINE__, "IIR21 resampler delay is bigger then MAX_DS_FILTER_DELAY");
writeOffset += IIR21_reSampler[0].delay*MAX_CHANNELS;
}
sampleRateAAC = getOutSampleRate();
config.bitRate = bitrate;
config.nChannelsIn=getInChannel();
config.nChannelsOut=nChannelsAAC;
config.bandWidth=bandwidth;
/* set up SBR configuration */
if(!IsSbrSettingAvail(bitrate, nChannelsAAC, sampleRateAAC, &sampleRateAAC))
throw Exception(__FILE__, __LINE__, "No valid SBR configuration found");
InitializeSbrDefaults (&sbrConfig);
sbrConfig.usePs = useParametricStereo;
AdjustSbrSettings( &sbrConfig,
bitrate,
nChannelsAAC,
sampleRateAAC,
AACENC_TRANS_FAC,
24000);
EnvOpen( &hEnvEnc,
inBuf + coreWriteOffset,
&sbrConfig,
&config.bandWidth);
/* set up AAC encoder, now that samling rate is known */
config.sampleRate = sampleRateAAC;
if (AacEncOpen(&aacEnc, config) != 0){
AacEncClose(aacEnc);
throw Exception(__FILE__, __LINE__, "Initialisation of AAC failed !");
}
init_plans();
/* create the ADTS header */
adts_hdr(outBuf, &config);
inSamples = AACENC_BLOCKSIZE * getInChannel() * 2;
// initialize the resampling coverter if needed
if ( converter ) {
#ifdef HAVE_SRC_LIB
converterData.input_frames = 4096/((getInBitsPerSample() / 8) * getInChannel());
converterData.data_in = new float[converterData.input_frames*getInChannel()];
converterData.output_frames = (int) (converterData.input_frames * resampleRatio + 1);
if ((int) inSamples > getInChannel() * converterData.output_frames) {
resampledOffset = new float[2 * inSamples];
} else {
resampledOffset = new float[2 * getInChannel() * converterData.input_frames];
}
converterData.src_ratio = resampleRatio;
converterData.end_of_input = 0;
#else
converter->initialize( resampleRatio, getInChannel());
//needed 2x(converted input samples) to handle offsets
int outCount = 2 * getInChannel() * (inSamples + 1);
if (resampleRatio > 1)
outCount = (int) (outCount * resampleRatio);
resampledOffset = new short int[outCount];
#endif
resampledOffsetSize = 0;
}
aacplusOpen = true;
reportEvent(10, "bitrate=", bitrate);
reportEvent(10, "nChannelsIn", getInChannel());
reportEvent(10, "nChannelsOut", getOutChannel());
reportEvent(10, "nChannelsSBR", nChannelsSBR);
reportEvent(10, "nChannelsAAC", nChannelsAAC);
reportEvent(10, "sampleRateAAC", sampleRateAAC);
reportEvent(10, "inSamples", inSamples);
return true;
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
aacPlusEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() || len == 0) {
return 0;
}
unsigned int channels = getInChannel();
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int sampleSize = (bitsPerSample / 8) * channels;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
unsigned int samples = (unsigned int) nSamples * channels;
int processedSamples = 0;
if ( converter ) {
unsigned int converted;
#ifdef HAVE_SRC_LIB
src_short_to_float_array ((short *) buf, converterData.data_in, samples);
converterData.input_frames = nSamples;
converterData.data_out = resampledOffset + (resampledOffsetSize * channels);
int srcError = src_process (converter, &converterData);
if (srcError)
throw Exception (__FILE__, __LINE__, "libsamplerate error: ", src_strerror (srcError));
converted = converterData.output_frames_gen;
#else
int inCount = nSamples;
short int * shortBuffer = new short int[samples];
int outCount = (int) (inCount * resampleRatio);
unsigned char * b = (unsigned char*) buf;
Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
converted = converter->resample( inCount,
outCount+1,
shortBuffer,
&resampledOffset[resampledOffsetSize*channels]);
delete[] shortBuffer;
#endif
resampledOffsetSize += converted;
// encode samples (if enough)
while(resampledOffsetSize - processedSamples >= inSamples/channels) {
#ifdef HAVE_SRC_LIB
short *shortData = new short[inSamples];
src_float_to_short_array(resampledOffset + (processedSamples * channels),
shortData, inSamples) ;
encodeAacSamples (shortData, inSamples, channels);
delete [] shortData;
#else
encodeAacSamples (&resampledOffset[processedSamples*channels], inSamples, channels);
#endif
processedSamples+=inSamples/channels;
}
if (processedSamples && (int) resampledOffsetSize >= processedSamples) {
resampledOffsetSize -= processedSamples;
//move least part of resampled data to beginning
if(resampledOffsetSize)
#ifdef HAVE_SRC_LIB
resampledOffset = (float *) memmove(resampledOffset, &resampledOffset[processedSamples*channels],
resampledOffsetSize*channels*sizeof(float));
#else
resampledOffset = (short *) memmove(resampledOffset, &resampledOffset[processedSamples*channels],
resampledOffsetSize*sampleSize);
#endif
}
} else {
encodeAacSamples ((short *) buf, samples, channels);
}
return samples;
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
LameLibEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() ) {
return 0;
}
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int inChannels = getInChannel();
unsigned int sampleSize = (bitsPerSample / 8) * inChannels;
unsigned char * b = (unsigned char*) buf;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
short int * leftBuffer = new short int[nSamples];
short int * rightBuffer = new short int[nSamples];
if ( bitsPerSample == 8 ) {
Util::conv8( b, processed, leftBuffer, rightBuffer, inChannels);
} else if ( bitsPerSample == 16 ) {
Util::conv16( b,
processed,
leftBuffer,
rightBuffer,
inChannels,
isInBigEndian());
} else {
delete[] leftBuffer;
delete[] rightBuffer;
throw Exception( __FILE__, __LINE__,
"unsupported number of bits per sample for the encoder",
bitsPerSample );
}
// data chunk size estimate according to lame documentation
// NOTE: mp3Size is calculated based on the number of input channels
// which may be bigger than need, as output channels can be less
unsigned int mp3Size = (unsigned int) (1.25 * nSamples + 7200);
unsigned char * mp3Buf = new unsigned char[mp3Size];
int ret;
ret = lame_encode_buffer( lameGlobalFlags,
leftBuffer,
inChannels == 2 ? rightBuffer : leftBuffer,
nSamples,
mp3Buf,
mp3Size );
delete[] leftBuffer;
delete[] rightBuffer;
if ( ret < 0 ) {
reportEvent( 3, "lame encoding error", ret);
delete[] mp3Buf;
return 0;
}
unsigned int written = sink->write( mp3Buf, ret);
delete[] mp3Buf;
// just let go data that could not be written
if ( written < (unsigned int) ret ) {
reportEvent( 2,
"couldn't write all from encoder to underlying sink",
ret - written);
}
return processed;
}
/*------------------------------------------------------------------------------
* Open an encoding session
*----------------------------------------------------------------------------*/
bool
FaacEncoder :: open ( void )
throw ( Exception )
{
if ( isOpen() ) {
close();
}
// open the underlying sink
if ( !getSink()->open() ) {
throw Exception( __FILE__, __LINE__,
"faac lib opening underlying sink error");
}
char * faacVersion;
char * faacCopyright;
faacEncGetVersion(&faacVersion, &faacCopyright);
reportEvent(1, "Using faac codec version", faacVersion);
encoderHandle = faacEncOpen(getOutSampleRate(),
getInChannel(),
&inputSamples,
&maxOutputBytes);
faacEncConfiguration * faacConfig;
faacConfig = faacEncGetCurrentConfiguration(encoderHandle);
faacConfig->aacObjectType = MAIN;
faacConfig->mpegVersion = MPEG2;
faacConfig->useTns = 1;
faacConfig->shortctl = SHORTCTL_NORMAL;
faacConfig->useLfe = 0;
faacConfig->allowMidside = 1;
faacConfig->bitRate = getOutBitrate() * 1000 / getOutChannel();
faacConfig->bandWidth = lowpass;
faacConfig->quantqual = (unsigned long) (getOutQuality() * 1000.0);
faacConfig->outputFormat = 1;
faacConfig->inputFormat = FAAC_INPUT_16BIT;
if (!faacEncSetConfiguration(encoderHandle, faacConfig)) {
throw Exception(__FILE__, __LINE__,
"error configuring faac library");
}
// initialize the resampling coverter if needed
if ( converter ) {
#ifdef HAVE_SRC_LIB
converterData.input_frames = 4096/((getInBitsPerSample() / 8) * getInChannel());
converterData.data_in = new float[converterData.input_frames*getInChannel()];
converterData.output_frames = (int) (converterData.input_frames * resampleRatio + 1);
if ((int) inputSamples > getInChannel() * converterData.output_frames) {
resampledOffset = new float[2 * inputSamples];
} else {
resampledOffset = new float[2 * getInChannel() * converterData.input_frames];
}
converterData.src_ratio = resampleRatio;
converterData.end_of_input = 0;
#else
converter->initialize( resampleRatio, getInChannel());
//needed 2x(converted input samples) to handle offsets
int outCount = 2 * getInChannel() * (inputSamples + 1);
if (resampleRatio > 1)
outCount = (int) (outCount * resampleRatio);
resampledOffset = new short int[outCount];
#endif
resampledOffsetSize = 0;
}
faacOpen = true;
return true;
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
LameLibEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() ) {
return 0;
}
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int channels = getInChannel();
if ( channels != 1 && channels != 2 ) {
throw Exception( __FILE__, __LINE__,
"unsupported number of channels for the encoder",
channels );
}
unsigned int sampleSize = (bitsPerSample / 8) * channels;
unsigned char * b = (unsigned char*) buf;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
short int leftBuffer[nSamples];
short int rightBuffer[nSamples];
if ( bitsPerSample == 8 ) {
conv8( b, processed, leftBuffer, rightBuffer, channels);
} else if ( bitsPerSample == 16 ) {
conv16( b, processed, leftBuffer, rightBuffer, channels);
} else {
throw Exception( __FILE__, __LINE__,
"unsupported number of bits per sample for the encoder",
bitsPerSample );
}
// data chunk size estimate according to lame documentation
unsigned int mp3Size = (unsigned int) (1.25 * nSamples + 7200);
unsigned char mp3Buf[mp3Size];
int ret;
ret = lame_encode_buffer( lameGlobalFlags,
leftBuffer,
channels == 2 ? rightBuffer : leftBuffer,
nSamples,
mp3Buf,
mp3Size );
if ( ret < 0 ) {
reportEvent( 3, "lame encoding error", ret);
return 0;
}
unsigned int written = sink->write( mp3Buf, ret);
// just let go data that could not be written
if ( written < (unsigned int) ret ) {
reportEvent( 2,
"couldn't write all from encoder to underlying sink",
ret - written);
}
return processed;
}
/*------------------------------------------------------------------------------
* Open an encoding session
*----------------------------------------------------------------------------*/
bool
aacPlusEncoder :: open ( void )
throw ( Exception )
{
if ( isOpen() ) {
close();
}
// open the underlying sink
if ( !sink->open() ) {
throw Exception( __FILE__, __LINE__,
"aacplus lib opening underlying sink error");
}
reportEvent(1, "Using aacplus codec");
encoderHandle = aacplusEncOpen(getOutSampleRate(),
getInChannel(),
&inputSamples,
&maxOutputBytes);
aacplusEncConfiguration * aacplusConfig;
aacplusConfig = aacplusEncGetCurrentConfiguration(encoderHandle);
aacplusConfig->bitRate = getOutBitrate() * 1000;
aacplusConfig->bandWidth = lowpass;
aacplusConfig->outputFormat = 1;
aacplusConfig->inputFormat = AACPLUS_INPUT_16BIT;
aacplusConfig->nChannelsOut = getOutChannel();
if (!aacplusEncSetConfiguration(encoderHandle, aacplusConfig)) {
throw Exception(__FILE__, __LINE__,
"error configuring libaacplus library");
}
// initialize the resampling coverter if needed
if ( converter ) {
#ifdef HAVE_SRC_LIB
converterData.input_frames = 4096/((getInBitsPerSample() / 8) * getInChannel());
converterData.data_in = new float[converterData.input_frames*getInChannel()];
converterData.output_frames = (int) (converterData.input_frames * resampleRatio + 1);
if ((int) inputSamples > getInChannel() * converterData.output_frames) {
resampledOffset = new float[2 * inputSamples];
} else {
resampledOffset = new float[2 * getInChannel() * converterData.input_frames];
}
converterData.src_ratio = resampleRatio;
converterData.end_of_input = 0;
#else
converter->initialize( resampleRatio, getInChannel());
//needed 2x(converted input samples) to handle offsets
int outCount = 2 * getInChannel() * (inputSamples + 1);
if (resampleRatio > 1)
outCount = (int) (outCount * resampleRatio);
resampledOffset = new short int[outCount];
#endif
resampledOffsetSize = 0;
}
aacplusOpen = true;
reportEvent(10, "nChannelsAAC", aacplusConfig->nChannelsOut);
reportEvent(10, "sampleRateAAC", aacplusConfig->sampleRate);
reportEvent(10, "inSamples", inputSamples);
return true;
}
/*------------------------------------------------------------------------------
* Initialize the encoder
*----------------------------------------------------------------------------*/
void
OpusLibEncoder :: init ( unsigned int outMaxBitrate )
throw ( Exception )
{
this->outMaxBitrate = outMaxBitrate;
if ( getInBitsPerSample() != 16 && getInBitsPerSample() != 8 ) {
throw Exception( __FILE__, __LINE__,
"specified bits per sample not supported",
getInBitsPerSample() );
}
if ( getInChannel() != 1 && getInChannel() != 2 ) {
throw Exception( __FILE__, __LINE__,
"unsupported number of channels for the encoder",
getInChannel() );
}
if ( getOutSampleRate() != 48000 ) {
throw Exception( __FILE__, __LINE__,
"unsupported sample rate for this encoder, you should resample your input to 48000Hz",
getOutSampleRate() );
}
if ( getOutSampleRate() == getInSampleRate() ) {
resampleRatio = 1;
converter = 0;
} else {
resampleRatio = ( (double) getOutSampleRate() /
(double) getInSampleRate() );
// Determine if we can use linear interpolation.
// The inverse of the ratio must be a power of two for linear mode to
// be of sufficient quality.
bool useLinear = true;
double inverse = 1 / resampleRatio;
int integer = (int) inverse;
// Check that the inverse of the ratio is an integer
if( integer == inverse ) {
while( useLinear && integer ) { // Loop through the bits
// If the lowest order bit is not the only one set
if( integer & 1 && integer != 1 ) {
// Not a power of two; cannot use linear
useLinear = false;
} else {
// Shift all the bits over and try again
integer >>= 1;
}
}
} else {
useLinear = false;
}
// If we get here and useLinear is still true, then we have
// a power of two.
// open the aflibConverter in
// - high quality
// - linear or quadratic (non-linear) based on algorithm
// - not filter interpolation
#ifdef HAVE_SRC_LIB
int srcError = 0;
converter = src_new(useLinear == true ? SRC_LINEAR : SRC_SINC_FASTEST,
getInChannel(), &srcError);
if(srcError)
throw Exception (__FILE__, __LINE__, "libsamplerate error: ", src_strerror (srcError));
#else
converter = new aflibConverter( true, useLinear, false);
#endif
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
aacPlusEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() || len == 0) {
return 0;
}
unsigned int channels = getInChannel();
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int sampleSize = (bitsPerSample / 8) * channels;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
unsigned int samples = (unsigned int) nSamples * channels;
unsigned int i;
int ch, outSamples, numOutBytes;
reportEvent(10, "converting short to float");
short *TimeDataPcm = (short *) buf;
if(channels == 2) {
for (i=0; i<samples; i++)
inBuf[i+writeOffset+writtenSamples] = (float) TimeDataPcm[i];
} else {
/* using only left channel buffer for mono encoder */
for (i=0; i<samples; i++)
inBuf[writeOffset+2*writtenSamples+2*i] = (float) TimeDataPcm[i];
}
writtenSamples+=samples;
reportEvent(10, "writtenSamples", writtenSamples);
if (writtenSamples < inSamples)
return samples;
/* encode one SBR frame */
reportEvent(10, "encode one SBR frame");
EnvEncodeFrame( hEnvEnc,
inBuf + envReadOffset,
inBuf + coreWriteOffset,
MAX_CHANNELS,
&numAncDataBytes,
ancDataBytes);
reportEvent(10, "numAncDataBytes=", numAncDataBytes);
/* 2:1 downsampling for AAC core */
if (!useParametricStereo) {
reportEvent(10, "2:1 downsampling for AAC core");
for( ch=0; ch<nChannelsAAC; ch++ )
IIR21_Downsample( &(IIR21_reSampler[ch]),
inBuf + writeOffset+ch,
writtenSamples/channels,
MAX_CHANNELS,
inBuf+ch,
&outSamples,
MAX_CHANNELS);
reportEvent(10, "outSamples=", outSamples);
}
/* encode one AAC frame */
reportEvent(10, "encode one AAC frame");
AacEncEncode( aacEnc,
inBuf,
useParametricStereo ? 1 : MAX_CHANNELS, /* stride (step) */
ancDataBytes,
&numAncDataBytes,
(unsigned *) (outBuf+ADTS_HEADER_SIZE),
&numOutBytes);
if (useParametricStereo) {
memcpy( inBuf,inBuf+AACENC_BLOCKSIZE,CORE_INPUT_OFFSET_PS*sizeof(float));
} else {
memmove( inBuf,inBuf+AACENC_BLOCKSIZE*2*MAX_CHANNELS,writeOffset*sizeof(float));
}
/* Write one frame of encoded audio */
if (numOutBytes) {
reportEvent(10, "Write one frame of encoded audio:", numOutBytes+ADTS_HEADER_SIZE);
adts_hdr_up(outBuf, numOutBytes);
sink->write(outBuf, numOutBytes+ADTS_HEADER_SIZE);
}
writtenSamples=0;
return samples;
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
aacPlusEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() || len == 0) {
return 0;
}
unsigned int channels = getInChannel();
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int sampleSize = (bitsPerSample / 8) * channels;
unsigned char * b = (unsigned char*) buf;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
unsigned char * aacplusBuf = new unsigned char[maxOutputBytes];
int samples = (int) nSamples * channels;
int processedSamples = 0;
if ( converter ) {
unsigned int converted;
#ifdef HAVE_SRC_LIB
src_short_to_float_array ((short *) b, converterData.data_in, samples);
converterData.input_frames = nSamples;
converterData.data_out = resampledOffset + (resampledOffsetSize * channels);
int srcError = src_process (converter, &converterData);
if (srcError)
throw Exception (__FILE__, __LINE__, "libsamplerate error: ", src_strerror (srcError));
converted = converterData.output_frames_gen;
#else
int inCount = nSamples;
short int * shortBuffer = new short int[samples];
int outCount = (int) (inCount * resampleRatio);
Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
converted = converter->resample( inCount,
outCount+1,
shortBuffer,
&resampledOffset[resampledOffsetSize*channels]);
delete[] shortBuffer;
#endif
resampledOffsetSize += converted;
// encode samples (if enough)
while(resampledOffsetSize - processedSamples >= inputSamples/channels) {
#ifdef HAVE_SRC_LIB
short *shortData = new short[inputSamples];
src_float_to_short_array(resampledOffset + (processedSamples * channels),
shortData, inputSamples) ;
int outputBytes = aacplusEncEncode(encoderHandle,
(int32_t*) shortData,
inputSamples,
aacplusBuf,
maxOutputBytes);
delete [] shortData;
#else
int outputBytes = aacplusEncEncode(encoderHandle,
(int32_t*) &resampledOffset[processedSamples*channels],
inputSamples,
aacplusBuf,
maxOutputBytes);
#endif
unsigned int wrote = getSink()->write(aacplusBuf, outputBytes);
if (wrote < outputBytes) {
reportEvent(3, "aacPlusEncoder :: write, couldn't write full data to underlying sink");
}
processedSamples+=inputSamples/channels;
}
if (processedSamples && (int) resampledOffsetSize >= processedSamples) {
resampledOffsetSize -= processedSamples;
//move least part of resampled data to beginning
if(resampledOffsetSize)
#ifdef HAVE_SRC_LIB
resampledOffset = (float *) memmove(resampledOffset, &resampledOffset[processedSamples*channels],
resampledOffsetSize*channels*sizeof(float));
#else
resampledOffset = (short *) memmove(resampledOffset, &resampledOffset[processedSamples*channels],
resampledOffsetSize*sampleSize);
#endif
}
} else {
while (processedSamples < samples) {
int inSamples = samples - processedSamples < (int) inputSamples
? samples - processedSamples
: inputSamples;
int outputBytes = aacplusEncEncode(encoderHandle,
(int32_t*) (b + processedSamples/sampleSize),
inSamples,
aacplusBuf,
maxOutputBytes);
unsigned int wrote = getSink()->write(aacplusBuf, outputBytes);
if (wrote < outputBytes) {
reportEvent(3, "aacPlusEncoder :: write, couldn't write full data to underlying sink");
}
processedSamples += inSamples;
}
}
delete[] aacplusBuf;
// return processedSamples;
return samples * sampleSize;
}