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/*------------------------------------------------------------------------------
Copyright (c) 2000-2007 Tyrell Corporation. All rights reserved.
Tyrell DarkIce
File : aacPlusEncoder.h
Version : $Revision$
Author : $Author$
Location : $HeadURL$
Copyright notice:
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
------------------------------------------------------------------------------*/
#ifndef AACP_ENCODER_H
#define AACP_ENCODER_H
#ifndef __cplusplus
#error This is a C++ include file
#endif
/* ============================================================ include files */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#ifdef HAVE_AACPLUS_LIB
#include <aacplus.h>
#else
#error configure with aacplus
#endif
#include <cassert>
#include <cstring>
#include "Ref.h"
#include "Exception.h"
#include "Reporter.h"
#include "AudioEncoder.h"
#include "Sink.h"
#ifdef HAVE_SRC_LIB
#include <samplerate.h>
#else
#include "aflibConverter.h"
#endif
/* ================================================================ constants */
/* =================================================================== macros */
/* =============================================================== data types */
/**
* A class representing aacplus AAC+ encoder.
*
* @author $Author$
* @version $Revision$
*/
class aacPlusEncoder : public AudioEncoder, public virtual Reporter
{
private:
/**
* 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
*/
Ref<Sink> sink;
/**
* The handle to the AAC+ encoder instance.
*/
aacplusEncHandle encoderHandle;
/**
* The maximum number of input samples to supply to the encoder.
*/
unsigned long inputSamples;
/**
* The maximum number of output bytes the encoder returns in one call.
*/
unsigned long maxOutputBytes;
/**
* Lowpass filter. Sound frequency in Hz, from where up the
* input is cut.
*/
int lowpass;
/**
* Initialize the object.
*
* @param sink the sink to send mp3 output to
* @exception Exception
*/
inline void
init ( Sink * sink, int lowpass) throw (Exception)
{
this->aacplusOpen = false;
this->sink = sink;
this->lowpass = lowpass;
/* TODO: if we have float as input, we don't need conversion */
if ( getInBitsPerSample() != 16 && getInBitsPerSample() != 32 ) {
throw Exception( __FILE__, __LINE__,
"specified bits per sample not supported",
getInBitsPerSample() );
}
if ( getInChannel() > 2 ) {
throw Exception( __FILE__, __LINE__,
"unsupported number of input channels for the encoder",
getInChannel() );
}
if ( getOutChannel() > 2 ) {
throw Exception( __FILE__, __LINE__,
"unsupported number of output channels for the encoder",
getOutChannel() );
}
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() );
}
}
/**
* De-initialize the object.
*
* @exception Exception
*/
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;
}
}
protected:
/**
* Default constructor. Always throws an Exception.
*
* @exception Exception
*/
inline
aacPlusEncoder ( void ) throw ( Exception )
{
throw Exception( __FILE__, __LINE__);
}
public:
/**
* Constructor.
*
* @param sink the sink to send mp3 output to
* @param inSampleRate sample rate of the input.
* @param inBitsPerSample number of bits per sample of the input.
* @param inChannel number of channels of the input.
* @param inBigEndian shows if the input is big or little endian
* @param outBitrateMode the bit rate mode of the output.
* @param outBitrate bit rate of the output (kbits/sec).
* @param outQuality the quality of the stream.
* @param outSampleRate sample rate of the output.
* If 0, inSampleRate is used.
* @param outChannel number of channels of the output.
* If 0, inChannel is used.
* @param lowpass frequency threshold for the lowpass filter.
* Input above this frequency is cut.
* If 0, aacplus's default values are used,
* which depends on the out sample rate.
* @exception Exception
*/
inline
aacPlusEncoder ( Sink * sink,
unsigned int inSampleRate,
unsigned int inBitsPerSample,
unsigned int inChannel,
bool inBigEndian,
BitrateMode outBitrateMode,
unsigned int outBitrate,
double outQuality,
unsigned int outSampleRate = 0,
unsigned int outChannel = 0,
int lowpass = 0)
throw ( Exception )
: AudioEncoder ( sink,
inSampleRate,
inBitsPerSample,
inChannel,
inBigEndian,
outBitrateMode,
outBitrate,
outQuality,
outSampleRate,
outChannel )
{
init( sink, lowpass);
}
/**
* Constructor.
*
* @param sink the sink to send mp3 output to
* @param as get input sample rate, bits per sample and channels
* from this AudioSource.
* @param outBitrateMode the bit rate mode of the output.
* @param outBitrate bit rate of the output (kbits/sec).
* @param outQuality the quality of the stream.
* @param outSampleRate sample rate of the output.
* If 0, input sample rate is used.
* @param outChannel number of channels of the output.
* If 0, input channel is used.
* @param lowpass frequency threshold for the lowpass filter.
* Input above this frequency is cut.
* If 0, aacplus's default values are used,
* which depends on the out sample rate.
* @exception Exception
*/
inline
aacPlusEncoder ( Sink * sink,
const AudioSource * as,
BitrateMode outBitrateMode,
unsigned int outBitrate,
double outQuality,
unsigned int outSampleRate = 0,
unsigned int outChannel = 0,
int lowpass = 0)
throw ( Exception )
: AudioEncoder ( sink,
as,
outBitrateMode,
outBitrate,
outQuality,
outSampleRate,
outChannel )
{
init( sink, lowpass );
}
/**
* Copy constructor.
*
* @param encoder the aacPlusEncoder to copy.
*/
inline
aacPlusEncoder ( const aacPlusEncoder & encoder )
throw ( Exception )
: AudioEncoder( encoder )
{
init( encoder.sink.get(), encoder.lowpass);
}
/**
* Destructor.
*
* @exception Exception
*/
inline virtual
~aacPlusEncoder ( void ) throw ( Exception )
{
if ( isOpen() ) {
close();
}
strip();
}
/**
* Assignment operator.
*
* @param encoder the aacPlusEncoder to assign this to.
* @return a reference to this aacPlusEncoder.
* @exception Exception
*/
inline virtual aacPlusEncoder &
operator= ( const aacPlusEncoder & encoder ) throw ( Exception )
{
if ( this != &encoder ) {
strip();
AudioEncoder::operator=( encoder);
init( encoder.sink.get(), encoder.lowpass);
}
return *this;
}
/**
* Get the version string of the underlying aacplus library.
*
* @return the version string of the underlying aacplus library.
*/
inline const char *
getAacPlusVersion( void )
{
char * id;
//char * copyright;
/* aacplusEncGetVersion(&id, ©right); */
return id;
}
/**
* Check whether encoding is in progress.
*
* @return true if encoding is in progress, false otherwise.
*/
inline virtual bool
isRunning ( void ) const throw ()
{
return isOpen();
}
/**
* Start encoding. This function returns as soon as possible,
* with encoding started in the background.
*
* @return true if encoding has started, false otherwise.
* @exception Exception
*/
inline virtual bool
start ( void ) throw ( Exception )
{
return open();
}
/**
* Stop encoding. Stops the encoding running in the background.
*
* @exception Exception
*/
inline virtual void
stop ( void ) throw ( Exception )
{
return close();
}
/**
* Open an encoding session.
*
* @return true if opening was successfull, false otherwise.
* @exception Exception
*/
virtual bool
open ( void ) throw ( Exception );
/**
* Check if the encoding session is open.
*
* @return true if the encoding session is open, false otherwise.
*/
inline virtual bool
isOpen ( void ) const throw ()
{
return aacplusOpen;
}
/**
* Check if the encoder is ready to accept data.
*
* @param sec the maximum seconds to block.
* @param usec micro seconds to block after the full seconds.
* @return true if the encoder is ready to accept data,
* false otherwise.
* @exception Exception
*/
inline virtual bool
canWrite ( unsigned int sec,
unsigned int usec ) throw ( Exception )
{
if ( !isOpen() ) {
return false;
}
return true;
}
/**
* Write data to the encoder.
* Buf is expected to be a sequence of big-endian 16 bit values,
* with left and right channels interleaved. Len is the number of
* bytes, must be a multiple of 4.
*
* @param buf the data to write.
* @param len number of bytes to write from buf.
* @return the number of bytes written (may be less than len).
* @exception Exception
*/
virtual unsigned int
write ( const void * buf,
unsigned int len ) throw ( Exception );
/**
* Flush all data that was written to the encoder to the underlying
* connection.
*
* @exception Exception
*/
virtual void
flush ( void ) throw ( Exception );
/**
* Close the encoding session.
*
* @exception Exception
*/
virtual void
close ( void ) throw ( Exception );
};
/* ================================================= external data structures */
/* ====================================================== function prototypes */
#endif /* AACP_ENCODER_H */
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