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Patch by Sergiy <piratfm@gmail.com>: implement sample rate conversion for all codecs using libsamplerate, and keeping internal aflibConverter as fallback

This commit is contained in:
rafael@riseup.net 2009-12-22 11:56:16 +00:00
parent 542ac4b022
commit 2fa04c0e34
8 changed files with 482 additions and 61 deletions
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38
darkice/trunk/configure.in
View File

@ -270,6 +270,7 @@ dnl link JACK sound server if requested
dnl-----------------------------------------------------------------------------
AC_SUBST(JACK_CFLAGS)
AC_SUBST(JACK_LDFLAGS)
AC_SUBST(JACK_INCFLAGS)
AC_ARG_WITH(jack,
[ --with-jack use JACK sound system [yes] ],
@ -301,6 +302,43 @@ else
fi
dnl-----------------------------------------------------------------------------
dnl link Secret Rabbit Code (aka libsamplerate) if requested
dnl-----------------------------------------------------------------------------
AC_SUBST(SRC_CFLAGS)
AC_SUBST(SRC_LDFLAGS)
AC_SUBST(SRC_INCFLAGS)
AC_ARG_WITH(samplerate,
[ --with-samplerate use Secret Rabbit Code (aka libsamplerate) for samplerate conversion [yes] ],
USE_SRC=${withval}, USE_SRC="yes" )
AC_ARG_WITH(samplerate-prefix,
[ --with-samplerate-prefix=DIR alternate location for samplerate [/usr]
look for libraries in SRC-PREFIX/lib,
for headers in SRC-PREFIX/include],
CONFIG_SRC_PREFIX="${withval}", CONFIG_SRC_PREFIX="/usr")
if test "x${USE_SRC}" = "xyes" ; then
AC_MSG_CHECKING( [for samplerate libraries at ${CONFIG_SRC_PREFIX}] )
LA_SEARCH_LIB( SRC_LIB_LOC, SRC_INC_LOC, libsamplerate.la libsamplerate.so libsamplerate.dylib, samplerate.h,
${CONFIG_SRC_PREFIX})
if test "x${SRC_LIB_LOC}" != "x" ; then
AC_DEFINE( HAVE_SRC_LIB, 1, [build with samplerate conversion through libsamplerate] )
if test "x${SRC_INC_LOC}" != "x${SYSTEM_INCLUDE}" ; then
SRC_INCFLAGS="-I${SRC_INC_LOC}"
fi
SRC_LDFLAGS="-L${SRC_LIB_LOC} -lsamplerate"
AC_MSG_RESULT( [found at ${CONFIG_SRC_PREFIX}] )
else
AC_MSG_WARN( [not found, building libsamplerate support])
fi
else
AC_MSG_RESULT( [building without libsamplerate support] )
fi
AM_CONDITIONAL(HAVE_SRC_LIB, test "x${SRC_LIB_LOC}" != "x")
dnl-----------------------------------------------------------------------------
dnl check for MSG_NOSIGNAL for the send() function in libsocket
dnl-----------------------------------------------------------------------------

117
darkice/trunk/src/FaacEncoder.cpp
View File

@ -81,7 +81,7 @@ FaacEncoder :: open ( void )
faacEncGetVersion(&faacVersion, &faacCopyright);
reportEvent(1, "Using faac codec version", faacVersion);
encoderHandle = faacEncOpen(getInSampleRate(),
encoderHandle = faacEncOpen(getOutSampleRate(),
getInChannel(),
&inputSamples,
&maxOutputBytes);
@ -107,6 +107,31 @@ FaacEncoder :: open ( void )
"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;
@ -134,25 +159,89 @@ FaacEncoder :: write ( const void * buf,
int samples = (int) nSamples * channels;
int processedSamples = 0;
while (processedSamples < samples) {
int outputBytes;
int inSamples = samples - processedSamples < (int) inputSamples
? samples - processedSamples
: inputSamples;
outputBytes = faacEncEncode(encoderHandle,
(int32_t*) (b + processedSamples/sampleSize),
inSamples,
faacBuf,
maxOutputBytes);
getSink()->write(faacBuf, outputBytes);
processedSamples += inSamples;
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) {
int outputBytes;
#ifdef HAVE_SRC_LIB
short *shortData = new short[inputSamples];
src_float_to_short_array(resampledOffset + (processedSamples * channels),
shortData, inputSamples) ;
outputBytes = faacEncEncode(encoderHandle,
(int32_t*) shortData,
inputSamples,
faacBuf,
maxOutputBytes);
delete [] shortData;
#else
outputBytes = faacEncEncode(encoderHandle,
(int32_t*) &resampledOffset[processedSamples*channels],
inputSamples,
faacBuf,
maxOutputBytes);
#endif
getSink()->write(faacBuf, outputBytes);
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 outputBytes;
int inSamples = samples - processedSamples < (int) inputSamples
? samples - processedSamples
: inputSamples;
outputBytes = faacEncEncode(encoderHandle,
(int32_t*) (b + processedSamples/sampleSize),
inSamples,
faacBuf,
maxOutputBytes);
getSink()->write(faacBuf, outputBytes);
processedSamples += inSamples;
}
}
delete[] faacBuf;
return processedSamples;
// return processedSamples;
return samples;
}

84
darkice/trunk/src/FaacEncoder.h
View File

@ -52,6 +52,11 @@
#include "Reporter.h"
#include "AudioEncoder.h"
#include "Sink.h"
#ifdef HAVE_SRC_LIB
#include <samplerate.h>
#else
#include "aflibConverter.h"
#endif
/* ================================================================ constants */
@ -98,6 +103,24 @@ class FaacEncoder : public AudioEncoder, public virtual Reporter
*/
int lowpass;
/**
* Resample ratio
*/
double resampleRatio;
/**
* sample rate converter object for possible resampling
*/
#ifdef HAVE_SRC_LIB
SRC_STATE *converter;
SRC_DATA converterData;
float *resampledOffset;
#else
aflibConverter *converter;
short *resampledOffset;
#endif
unsigned int resampledOffsetSize;
/**
* Initialize the object.
*
@ -133,6 +156,58 @@ class FaacEncoder : public AudioEncoder, public virtual Reporter
throw Exception( __FILE__, __LINE__,
"input channels and output channels do not match");
}
if ( getOutSampleRate() == getInSampleRate() ) {
resampleRatio = 1;
converter = 0;
} else if (getInBitsPerSample() == 16) {
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
} else {
throw Exception( __FILE__, __LINE__,
"specified bits per sample with samplerate conversion not supported",
getInBitsPerSample() );
}
}
/**
@ -143,6 +218,15 @@ class FaacEncoder : public AudioEncoder, public virtual Reporter
inline void
strip ( void ) throw ( Exception )
{
if ( converter ) {
#ifdef HAVE_SRC_LIB
delete [] converterData.data_in;
src_delete (converter);
#else
delete converter;
#endif
delete [] resampledOffset;
}
}

35
darkice/trunk/src/Makefile.am
View File

@ -1,9 +1,21 @@
bin_PROGRAMS = darkice
AM_CXXFLAGS = -O2 -pedantic -Wall @DEBUG_CXXFLAGS@ @PTHREAD_CFLAGS@
@JACK_CFLAGS@
INCLUDES = @LAME_INCFLAGS@ @VORBIS_INCFLAGS@ @FAAC_INCFLAGS@ @AACPLUS_INCFLAGS@ @TWOLAME_INCFLAGS@ @ALSA_INCFLAGS@
INCLUDES = @LAME_INCFLAGS@ @VORBIS_INCFLAGS@ @FAAC_INCFLAGS@ @AACPLUS_INCFLAGS@ @TWOLAME_INCFLAGS@ \
@ALSA_INCFLAGS@ @JACK_INCFLAGS@ @SRC_INCFLAGS@
LDADD = @PTHREAD_LIBS@ @LAME_LDFLAGS@ @VORBIS_LDFLAGS@ @FAAC_LDFLAGS@ @AACPLUS_LDFLAGS@ @TWOLAME_LDFLAGS@ \
@ALSA_LDFLAGS@ @JACK_LDFLAGS@
@ALSA_LDFLAGS@ @JACK_LDFLAGS@ @SRC_LDFLAGS@
if HAVE_SRC_LIB
AFLIB_SOURCE =
else
AFLIB_SOURCE = aflibDebug.h\
aflibDebug.cc\
aflibConverter.h\
aflibConverter.cc\
aflibConverterLargeFilter.h\
aflibConverterSmallFilter.h
endif
darkice_SOURCES = AudioEncoder.h\
AudioSource.h\
@ -40,12 +52,6 @@ darkice_SOURCES = AudioEncoder.h\
FaacEncoder.h\
aacPlusEncoder.cpp\
aacPlusEncoder.h\
aflibDebug.h\
aflibDebug.cc\
aflibConverter.h\
aflibConverter.cc\
aflibConverterLargeFilter.h\
aflibConverterSmallFilter.h\
OssDspSource.cpp\
OssDspSource.h\
SerialUlaw.cpp\
@ -68,7 +74,14 @@ darkice_SOURCES = AudioEncoder.h\
Reporter.cpp\
AlsaDspSource.h\
AlsaDspSource.cpp\
JackDspSource.h\
JackDspSource.cpp\
main.cpp
JackDspSource.h\
JackDspSource.cpp\
main.cpp \
$(AFLIB_SOURCE)
EXTRA_darkice_SOURCES = aflibDebug.h\
aflibDebug.cc\
aflibConverter.h\
aflibConverter.cc\
aflibConverterLargeFilter.h\
aflibConverterSmallFilter.h

30
darkice/trunk/src/VorbisLibEncoder.cpp
View File

@ -117,7 +117,15 @@ VorbisLibEncoder :: init ( unsigned int outMaxBitrate )
// - 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
}
encoderOpen = false;
@ -243,7 +251,16 @@ VorbisLibEncoder :: open ( void )
// 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);
converterData.data_out = new float[getInChannel() * converterData.output_frames];
converterData.src_ratio = resampleRatio;
converterData.end_of_input = 0;
#else
converter->initialize( resampleRatio, getInChannel());
#endif
}
encoderOpen = true;
@ -307,13 +324,24 @@ VorbisLibEncoder :: write ( const void * buf,
// resample if needed
int inCount = nSamples;
int outCount = (int) (inCount * resampleRatio);
short int * resampledBuffer = new short int[outCount * channels];
short int * resampledBuffer = new short int[(outCount+1)* channels];
int converted;
#ifdef HAVE_SRC_LIB
converterData.input_frames = nSamples;
src_short_to_float_array (shortBuffer, converterData.data_in, totalSamples);
int srcError = src_process (converter, &converterData);
if (srcError)
throw Exception (__FILE__, __LINE__, "libsamplerate error: ", src_strerror (srcError));
converted = converterData.output_frames_gen;
src_float_to_short_array(converterData.data_out, resampledBuffer, converted*channels);
#else
converted = converter->resample( inCount,
outCount,
shortBuffer,
resampledBuffer );
#endif
vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState,
converted);

17
darkice/trunk/src/VorbisLibEncoder.h
View File

@ -52,7 +52,11 @@
#include "Reporter.h"
#include "AudioEncoder.h"
#include "CastSink.h"
#ifdef HAVE_SRC_LIB
#include <samplerate.h>
#else
#include "aflibConverter.h"
#endif
/* ================================================================ constants */
@ -115,9 +119,14 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
double resampleRatio;
/**
* aflibConverter object for possible resampling
* sample rate converter object for possible resampling
*/
#ifdef HAVE_SRC_LIB
SRC_STATE * converter;
SRC_DATA converterData;
#else
aflibConverter * converter;
#endif
/**
* Initialize the object.
@ -137,7 +146,13 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
strip ( void ) throw ( Exception )
{
if ( converter ) {
#ifdef HAVE_SRC_LIB
delete [] converterData.data_in;
delete [] converterData.data_out;
src_delete (converter);
#else
delete converter;
#endif
}
}

127
darkice/trunk/src/aacPlusEncoder.cpp
View File

@ -113,7 +113,7 @@ aacPlusEncoder :: open ( void )
writeOffset += IIR21_reSampler[0].delay*MAX_CHANNELS;
}
sampleRateAAC = getInSampleRate();
sampleRateAAC = getOutSampleRate();
config.bitRate = bitrate;
config.nChannelsIn=getInChannel();
config.nChannelsOut=nChannelsAAC;
@ -152,6 +152,31 @@ aacPlusEncoder :: open ( void )
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());
@ -181,34 +206,84 @@ aacPlusEncoder :: write ( const void * buf,
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
unsigned int samples = (unsigned int) nSamples * channels;
int processedSamples = 0;
unsigned int i;
int ch, outSamples, numOutBytes;
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;
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];
// 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 {
/* using only left channel buffer for mono encoder */
for (i=0; i<samples; i++)
inBuf[writeOffset+2*writtenSamples+2*i] = (float) TimeDataPcm[i];
encodeAacSamples ((short *) buf, samples, channels);
}
return samples;
}
void
aacPlusEncoder :: encodeAacSamples (short *TimeDataPcm, unsigned int samples, int channels)
throw ( Exception )
{
unsigned int i;
int ch, outSamples, numOutBytes;
for (i=0; i<samples; i++)
inBuf[(2/channels)*i+writeOffset+writtenSamples] = (float) TimeDataPcm[i];
writtenSamples+=samples;
reportEvent(10, "writtenSamples", writtenSamples);
if (writtenSamples < inSamples)
return samples;
return;
/* encode one SBR frame */
reportEvent(10, "encode one SBR frame");
EnvEncodeFrame( hEnvEnc,
inBuf + envReadOffset,
inBuf + coreWriteOffset,
@ -216,11 +291,8 @@ aacPlusEncoder :: write ( const void * buf,
&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,
@ -229,12 +301,9 @@ aacPlusEncoder :: write ( const void * buf,
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) */
@ -250,16 +319,14 @@ aacPlusEncoder :: write ( const void * buf,
/* 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);
adts_hdr_up(outBuf, numOutBytes);
sink->write(outBuf, numOutBytes+ADTS_HEADER_SIZE);
}
writtenSamples=0;
return samples;
}
return;
}
/*------------------------------------------------------------------------------
* Flush the data from the encoder

95
darkice/trunk/src/aacPlusEncoder.h
View File

@ -65,6 +65,11 @@ extern "C" {
#include "Reporter.h"
#include "AudioEncoder.h"
#include "Sink.h"
#ifdef HAVE_SRC_LIB
#include <samplerate.h>
#else
#include "aflibConverter.h"
#endif
/* ================================================================ constants */
@ -95,7 +100,25 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
* A flag to indicate if the encoding session is open.
*/
bool aacplusOpen;
/**
* Resample ratio
*/
double resampleRatio;
/**
* sample rate converter object for possible resampling
*/
#ifdef HAVE_SRC_LIB
SRC_STATE *converter;
SRC_DATA converterData;
float *resampledOffset;
#else
aflibConverter *converter;
short *resampledOffset;
#endif
unsigned int resampledOffsetSize;
/**
* The Sink to dump aac+ data to
*/
@ -161,7 +184,60 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
throw Exception( __FILE__, __LINE__,
"input channels and output channels do not match");
}
if ( getOutSampleRate() == getInSampleRate() ) {
resampleRatio = 1;
converter = 0;
} else if (getInBitsPerSample() == 16) {
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
} else {
throw Exception( __FILE__, __LINE__,
"specified bits per sample with samplerate conversion not supported",
getInBitsPerSample() );
}
bitrate = getOutBitrate() * 1000;
bandwidth = 0;
useParametricStereo = 0;
@ -171,8 +247,7 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
writeOffset = INPUT_DELAY*MAX_CHANNELS;
writtenSamples = 0;
aacEnc = NULL;
hEnvEnc=NULL;
hEnvEnc=NULL;
}
/**
@ -183,8 +258,20 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
inline void
strip ( void ) throw ( Exception )
{
if ( converter ) {
#ifdef HAVE_SRC_LIB
delete [] converterData.data_in;
src_delete (converter);
#else
delete converter;
#endif
delete [] resampledOffset;
}
}
void
encodeAacSamples (short *TimeDataPcm, unsigned int samples, int channels)
throw ( Exception );
protected: