/*------------------------------------------------------------------------------
* 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);
}
/*------------------------------------------------------------------------------
* Open an encoding session
*----------------------------------------------------------------------------*/
bool
LameLibEncoder :: open ( void )
throw ( Exception )
{
if ( isOpen() ) {
close();
}
lameGlobalFlags = lame_init();
// ugly lame returns -1 in a pointer on allocation errors
if ( !lameGlobalFlags || ((int)lameGlobalFlags) == -1 ) {
throw Exception( __FILE__, __LINE__,
"lame lib init error",
(int) lameGlobalFlags);
}
if ( 0 > lame_set_num_channels( lameGlobalFlags, getInChannel()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting channels error",
getInChannel() );
}
if ( 0 > lame_set_mode( lameGlobalFlags,
getOutChannel() == 1 ? MONO : JOINT_STEREO) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting mode error",
JOINT_STEREO );
}
reportEvent( 5, "set lame mode", lame_get_mode( lameGlobalFlags));
reportEvent( 5,
"set lame channels",
lame_get_num_channels( lameGlobalFlags));
if ( 0 > lame_set_in_samplerate( lameGlobalFlags, getInSampleRate()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting input sample rate error",
getInSampleRate() );
}
reportEvent( 5,
"set lame in sample rate",
lame_get_in_samplerate( lameGlobalFlags));
if ( 0 > lame_set_out_samplerate( lameGlobalFlags, getOutSampleRate()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting output sample rate error",
getOutSampleRate() );
}
reportEvent( 5,
"set lame out sample rate",
lame_get_out_samplerate( lameGlobalFlags));
switch ( getOutBitrateMode() ) {
case cbr: {
if ( 0 > lame_set_brate( lameGlobalFlags, getOutBitrate()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting output bit rate error",
getOutBitrate() );
}
reportEvent( 5,
"set lame bit rate",
lame_get_brate( lameGlobalFlags));
double d = (1.0 - getOutQuality()) * 10.0;
int q = int (d + 0.499999);
if ( 0 > lame_set_quality( lameGlobalFlags, q) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting quality error", q);
}
reportEvent( 5,
"set lame quality",
lame_get_quality( lameGlobalFlags));
} break;
case abr:
if ( 0 > lame_set_VBR( lameGlobalFlags,vbr_abr)) {
throw Exception( __FILE__, __LINE__,
"lame lib setting abr error", vbr_abr);
}
reportEvent( 5,
"set lame abr bitrate",
lame_get_VBR( lameGlobalFlags));
if ( 0 > lame_set_VBR_mean_bitrate_kbps( lameGlobalFlags,
getOutBitrate())) {
throw Exception( __FILE__, __LINE__,
"lame lib setting abr mean bitrate error",
getOutBitrate());
}
reportEvent( 5,
"set lame abr mean bitrate",
lame_get_VBR_mean_bitrate_kbps( lameGlobalFlags));
break;
case vbr: {
if ( 0 > lame_set_VBR( lameGlobalFlags, vbr_mtrh)) {
throw Exception( __FILE__, __LINE__,
"lame lib setting vbr error", vbr_mtrh );
}
reportEvent( 5,
"set lame vbr bitrate",
lame_get_VBR( lameGlobalFlags));
double d = (1.0 - getOutQuality()) * 10.0;
int q = int (d + 0.499999);
if ( 0 > lame_set_VBR_q( lameGlobalFlags, q) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting vbr quality error", q);
}
reportEvent( 5,
"set lame vbr quality",
lame_get_VBR_q( lameGlobalFlags));
} break;
}
if ( 0 > lame_set_lowpassfreq( lameGlobalFlags, lowpass) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting lowpass frequency error",
lowpass );
}
reportEvent( 5,
"set lame lowpass frequency",
lame_get_lowpassfreq( lameGlobalFlags));
if ( 0 > lame_set_highpassfreq( lameGlobalFlags, highpass) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting highpass frequency error",
lowpass );
}
reportEvent( 5,
"set lame highpass frequency",
lame_get_highpassfreq( lameGlobalFlags));
// not configurable lame settings
if ( 0 > lame_set_exp_nspsytune( lameGlobalFlags, 1) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting psycho acoustic model error");
}
reportEvent( 5,
"set lame psycho acoustic model",
lame_get_exp_nspsytune( lameGlobalFlags));
if ( 0 > lame_set_error_protection( lameGlobalFlags, 1) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting error protection error",
1 );
}
reportEvent( 5,
"set lame error protection",
lame_get_error_protection( lameGlobalFlags));
// let lame init its own params based on our settings
if ( 0 > lame_init_params( lameGlobalFlags) ) {
throw Exception( __FILE__, __LINE__,
"lame lib initializing params error" );
}
lame_print_config( lameGlobalFlags);
// open the underlying sink
if ( !sink->open() ) {
throw Exception( __FILE__, __LINE__,
"lame lib opening underlying sink error");
}
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
}
/*------------------------------------------------------------------------------
* Open an encoding session
*----------------------------------------------------------------------------*/
bool
LameLibEncoder :: open ( void )
throw ( Exception )
{
if ( isOpen() ) {
close();
}
lameGlobalFlags = lame_init();
// ugly lame returns -1 in a pointer on allocation errors
if ( !lameGlobalFlags || ((int)lameGlobalFlags) == -1 ) {
throw Exception( __FILE__, __LINE__,
"lame lib init error",
(int) lameGlobalFlags);
}
if ( 0 > lame_set_num_channels( lameGlobalFlags, getInChannel()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting channels error",
getInChannel() );
}
if ( 0 > lame_set_mode( lameGlobalFlags,
getInChannel() == 1 ? MONO : JOINT_STEREO) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting mode error",
JOINT_STEREO );
}
reportEvent( 5, "set lame mode", lame_get_mode( lameGlobalFlags));
reportEvent( 5,
"set lame channels",
lame_get_num_channels( lameGlobalFlags));
if ( 0 > lame_set_in_samplerate( lameGlobalFlags, getInSampleRate()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting input sample rate error",
getInSampleRate() );
}
reportEvent( 5,
"set lame in sample rate",
lame_get_in_samplerate( lameGlobalFlags));
if ( 0 > lame_set_out_samplerate( lameGlobalFlags, getOutSampleRate()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting output sample rate error",
getOutSampleRate() );
}
reportEvent( 5,
"set lame out sample rate",
lame_get_out_samplerate( lameGlobalFlags));
if ( 0 > lame_set_brate( lameGlobalFlags, getOutBitrate()) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting output bit rate error",
getOutBitrate() );
}
reportEvent( 5, "set lame bit rate", lame_get_brate( lameGlobalFlags));
if ( lowpass ) {
if ( 0 > lame_set_lowpassfreq( lameGlobalFlags, lowpass) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting lowpass frequency error",
lowpass );
}
reportEvent( 5,
"set lame lowpass frequency",
lame_get_lowpassfreq( lameGlobalFlags));
}
if ( highpass ) {
if ( 0 > lame_set_highpassfreq( lameGlobalFlags, highpass) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting highpass frequency error",
lowpass );
}
reportEvent( 5,
"set lame highpass frequency",
lame_get_highpassfreq( lameGlobalFlags));
}
// not configurable lame settings
if ( 0 > lame_set_quality( lameGlobalFlags, 2) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting quality error",
2 );
}
reportEvent( 5, "set lame quality", lame_get_quality( lameGlobalFlags));
if ( 0 > lame_set_exp_nspsytune( lameGlobalFlags, 1) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting psycho acoustic model error");
}
reportEvent( 5,
"set lame psycho acoustic model",
lame_get_exp_nspsytune( lameGlobalFlags));
if ( 0 > lame_set_error_protection( lameGlobalFlags, 1) ) {
throw Exception( __FILE__, __LINE__,
"lame lib setting error protection error",
1 );
}
reportEvent( 5,
"set lame error protection",
lame_get_error_protection( lameGlobalFlags));
// let lame init its own params based on our settings
if ( 0 > lame_init_params( lameGlobalFlags) ) {
throw Exception( __FILE__, __LINE__,
"lame lib initializing params error" );
}
lame_print_config( lameGlobalFlags);
// open the underlying sink
if ( !sink->open() ) {
throw Exception( __FILE__, __LINE__,
"lame lib opening underlying sink error");
}
return true;
}
/*------------------------------------------------------------------------------
* 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 = getInSampleRate();
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;
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;
}
