diff --git a/darkice/trunk/src/VorbisLibEncoder.cpp b/darkice/trunk/src/VorbisLibEncoder.cpp index eca8ab6..77d67db 100644 --- a/darkice/trunk/src/VorbisLibEncoder.cpp +++ b/darkice/trunk/src/VorbisLibEncoder.cpp @@ -72,13 +72,38 @@ VorbisLibEncoder :: open ( void ) } vorbis_info_init( &vorbisInfo); - if ( (ret = vorbis_encode_init( &vorbisInfo, - getInChannel(), - getInSampleRate(), - getOutBitrate() * 1000, - getOutBitrate() * 1000, - -1 )) ) { - throw Exception( __FILE__, __LINE__, "vorbis encode init error", ret); + + if ( isVBR() ) { + + if ( (ret = vorbis_encode_init_vbr( &vorbisInfo, + getInChannel(), + getOutSampleRate(), + getOutQuality() )) ) { + throw Exception( __FILE__, __LINE__, + "vorbis encode init error", ret); + } + } else { +#ifdef VORBIS_LIB_RC3 + if ( (ret = vorbis_encode_init( &vorbisInfo, + getInChannel(), + getOutSampleRate(), + getOutBitrate() * 1024, + getOutBitrate() * 1024, + -1 )) ) { + throw Exception( __FILE__, __LINE__, + "vorbis encode init error", ret); + } +#else + if ( (ret = vorbis_encode_init( &vorbisInfo, + getInChannel(), + getOutSampleRate(), + -1, + getOutBitrate() * 1024, + -1 )) ) { + throw Exception( __FILE__, __LINE__, + "vorbis encode init error", ret); + } +#endif } if ( (ret = vorbis_analysis_init( &vorbisDspState, &vorbisInfo)) ) { @@ -100,8 +125,6 @@ VorbisLibEncoder :: open ( void ) } // create an empty vorbis_comment structure - vorbis_comment vorbisComment; - vorbis_comment_init( &vorbisComment); // create the vorbis stream headers and send them to the underlying sink @@ -117,8 +140,6 @@ VorbisLibEncoder :: open ( void ) throw Exception( __FILE__, __LINE__, "vorbis header init error", ret); } - vorbis_comment_init( &vorbisComment); - ogg_stream_packetin( &oggStreamState, &header); ogg_stream_packetin( &oggStreamState, &commentHeader); ogg_stream_packetin( &oggStreamState, &codeHeader); @@ -131,116 +152,16 @@ VorbisLibEncoder :: open ( void ) vorbis_comment_clear( &vorbisComment ); + // initialize the resampling coverter if needed + if ( converter ) { + converter->initialize( resampleRatio, getInChannel()); + } + encoderOpen = true; return true; } -/*------------------------------------------------------------------------------ - * Convert an unsigned char buffer holding 8 bit PCM values with channels - * interleaved to two float buffers (one for each channel) - *----------------------------------------------------------------------------*/ -void -VorbisLibEncoder :: conv8 ( unsigned char * pcmBuffer, - unsigned int lenPcmBuffer, - float * leftBuffer, - float * rightBuffer, - unsigned int channels ) -{ - if ( channels == 1 ) { - unsigned int i, j; - - for ( i = 0, j = 0; i < lenPcmBuffer; ) { - short int value; - - value = pcmBuffer[i++]; - leftBuffer[j] = ((float) value) / 128.f; - ++j; - } - } else { - unsigned int i, j; - - for ( i = 0, j = 0; i < lenPcmBuffer; ) { - short int value; - - value = pcmBuffer[i++]; - leftBuffer[j] = ((float) value) / 128.f; - value = pcmBuffer[i++]; - rightBuffer[j] = ((float) value) / 128.f; - ++j; - } - } -} - - -/*------------------------------------------------------------------------------ - * Convert an unsigned char buffer holding 16 bit PCM values with channels - * interleaved to two float buffers (one for each channel) - *----------------------------------------------------------------------------*/ -void -VorbisLibEncoder :: conv16 ( unsigned char * pcmBuffer, - unsigned int lenPcmBuffer, - float * leftBuffer, - float * rightBuffer, - unsigned int channels ) -{ - if ( isInBigEndian() ) { - if ( channels == 1 ) { - unsigned int i, j; - - for ( i = 0, j = 0; i < lenPcmBuffer; ) { - short int value; - - value = pcmBuffer[i++] << 8; - value |= pcmBuffer[i++]; - leftBuffer[j] = ((float) value) / 32768.f; - ++j; - } - } else { - unsigned int i, j; - - for ( i = 0, j = 0; i < lenPcmBuffer; ) { - short int value; - - value = pcmBuffer[i++] << 8; - value |= pcmBuffer[i++]; - leftBuffer[j] = ((float) value) / 32768.f; - value = pcmBuffer[i++] << 8; - value |= pcmBuffer[i++]; - rightBuffer[j] = ((float) value) / 32768.f; - ++j; - } - } - } else { - if ( channels == 1 ) { - unsigned int i, j; - - for ( i = 0, j = 0; i < lenPcmBuffer; ) { - short int value; - - value = pcmBuffer[i++]; - value |= pcmBuffer[i++] << 8; - leftBuffer[j] = ((float) value) / 32768.f; - ++j; - } - } else { - unsigned int i, j; - - for ( i = 0, j = 0; i < lenPcmBuffer; ) { - short int value; - - value = pcmBuffer[i++]; - value |= pcmBuffer[i++] << 8; - leftBuffer[j] = ((float) value) / 32768.f; - value = pcmBuffer[i++]; - value |= pcmBuffer[i++] << 8; - rightBuffer[j] = ((float) value) / 32768.f; - ++j; - } - } - } -} - /*------------------------------------------------------------------------------ * Write data to the encoder @@ -268,19 +189,37 @@ VorbisLibEncoder :: write ( const void * buf, unsigned int nSamples = processed / sampleSize; float ** vorbisBuffer; - vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState, nSamples); - if ( bitsPerSample == 8 ) { - conv8( b, processed, vorbisBuffer[0], vorbisBuffer[1], channels); - } else if ( bitsPerSample == 16 ) { - conv16( b, processed, vorbisBuffer[0], vorbisBuffer[1], channels); + // 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 { - throw Exception( __FILE__, __LINE__, - "unsupported number of bits per sample for the encoder", - bitsPerSample ); + + vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState, nSamples); + Util::conv( shortBuffer, totalSamples, vorbisBuffer, channels); + vorbis_analysis_wrote( &vorbisDspState, nSamples); } - vorbis_analysis_wrote( &vorbisDspState, nSamples); vorbisBlocksOut(); return processed; @@ -315,9 +254,11 @@ VorbisLibEncoder :: vorbisBlocksOut ( void ) throw () ogg_page oggPage; vorbis_analysis( &vorbisBlock, &oggPacket); +#ifdef VORBIS_LIB_RC3 vorbis_bitrate_addblock( &vorbisBlock); while ( vorbis_bitrate_flushpacket( &vorbisDspState, &oggPacket) ) { +#endif ogg_stream_packetin( &oggStreamState, &oggPacket); @@ -334,7 +275,9 @@ VorbisLibEncoder :: vorbisBlocksOut ( void ) throw () oggPage.header_len + oggPage.body_len - written); } } +#ifdef VORBIS_LIB_RC3 } +#endif } } @@ -351,6 +294,7 @@ VorbisLibEncoder :: close ( void ) throw ( Exception ) ogg_stream_clear( &oggStreamState); vorbis_block_clear( &vorbisBlock); vorbis_dsp_clear( &vorbisDspState); + vorbis_comment_clear( &vorbisComment); vorbis_info_clear( &vorbisInfo); encoderOpen = false; @@ -366,6 +310,11 @@ VorbisLibEncoder :: close ( void ) throw ( Exception ) $Source$ $Log$ + Revision 1.7 2002/03/28 16:47:38 darkeye + moved functions conv8() and conv16() to class Util (as conv()) + added resampling functionality + added support for variable bitrates + Revision 1.6 2002/02/20 10:35:35 darkeye updated to work with Ogg Vorbis libs rc3 and current IceCast2 cvs diff --git a/darkice/trunk/src/VorbisLibEncoder.h b/darkice/trunk/src/VorbisLibEncoder.h index 5308f30..1e7a53a 100644 --- a/darkice/trunk/src/VorbisLibEncoder.h +++ b/darkice/trunk/src/VorbisLibEncoder.h @@ -52,6 +52,7 @@ #include "Reporter.h" #include "AudioEncoder.h" #include "Sink.h" +#include "aflibConverter.h" /* ================================================================ constants */ @@ -93,6 +94,11 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter */ vorbis_block vorbisBlock; + /** + * Ogg Vorbis library global comment + */ + vorbis_comment vorbisComment; + /** * Ogg library global stream state */ @@ -103,6 +109,16 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter */ Ref sink; + /** + * Resample ratio + */ + double resampleRatio; + + /** + * aflibConverter object for possible resampling + */ + aflibConverter * converter; + /** * Initialize the object. * @@ -120,9 +136,17 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter getInBitsPerSample() ); } - if ( getOutSampleRate() != getInSampleRate() ) { - throw Exception( __FILE__, __LINE__, - "different in and out sample rate not supported"); + if ( getOutSampleRate() == getInSampleRate() ) { + resampleRatio = 1; + converter = 0; + } else { + resampleRatio = ( (double) getOutSampleRate() / + (double) getInSampleRate() ); + // open the aflibConverter in + // - high quality + // - not linear (quadratic) interpolation + // - not filter interpolation + converter = new aflibConverter( true, true, false); } if ( getInChannel() != getOutChannel() ) { @@ -143,42 +167,6 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter { } - /** - * Convert a char buffer holding 8 bit PCM values to a short buffer - * - * @param pcmBuffer buffer holding 8 bit PCM audio values, - * channels are interleaved - * @param lenPcmBuffer length of pcmBuffer - * @param leftBuffer put the left channel here (must be big enough) - * @param rightBuffer put the right channel here (if mono, not - * touched, must be big enough) - * @param channels number of channels (1 = mono, 2 = stereo) - */ - void - conv8 ( unsigned char * pcmBuffer, - unsigned int lenPcmBuffer, - float * leftBuffer, - float * rightBuffer, - unsigned int channels ); - - /** - * Convert a char buffer holding 16 bit PCM values to a short buffer - * - * @param pcmBuffer buffer holding 16 bit PCM audio values, - * channels are interleaved - * @param lenPcmBuffer length of pcmBuffer - * @param leftBuffer put the left channel here (must be big enough) - * @param rightBuffer put the right channel here (if mono, not - * touched, must be big enough) - * @param channels number of channels (1 = mono, 2 = stereo) - */ - void - conv16 ( unsigned char * pcmBuffer, - unsigned int lenPcmBuffer, - float * leftBuffer, - float * rightBuffer, - unsigned int channels ); - /** * Send pending Vorbis blocks to the underlying stream */ @@ -203,13 +191,13 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter public: /** - * Constructor. + * Constructor for fixed bitrate encoding. * * @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 outBitrate bit rate of the output (bits/sec). + * @param outBitrate bit rate of the output (kbits/sec). * @param outSampleRate sample rate of the output. * If 0, inSampleRate is used. * @param outChannel number of channels of the output. @@ -237,12 +225,46 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter } /** - * Constructor. + * Constructor for variable bitrate encoding. + * + * @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 outQuality the quality of the stream (0.0 .. 1.0). + * @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. + * @exception Exception + */ + inline + VorbisLibEncoder ( Sink * sink, + unsigned int inSampleRate, + unsigned int inBitsPerSample, + unsigned int inChannel, + double outQuality, + unsigned int outSampleRate = 0, + unsigned int outChannel = 0 ) + throw ( Exception ) + + : AudioEncoder ( inSampleRate, + inBitsPerSample, + inChannel, + outQuality, + outSampleRate, + outChannel ) + { + init( sink); + } + + /** + * Constructor for fixed bitrate encoding. * * @param sink the sink to send mp3 output to * @param as get input sample rate, bits per sample and channels * from this AudioSource. - * @param outBitrate bit rate of the output (bits/sec). + * @param outBitrate bit rate of the output (kbits/sec). * @param outSampleRate sample rate of the output. * If 0, input sample rate is used. * @param outChannel number of channels of the output. @@ -265,6 +287,35 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter init( sink); } + /** + * Constructor for variable bitrate encoding. + * + * @param sink the sink to send mp3 output to + * @param as get input sample rate, bits per sample and channels + * from this AudioSource. + * @param outQuality the quality of the stream (0.0 .. 1.0). + * @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. + * @exception Exception + */ + inline + VorbisLibEncoder ( Sink * sink, + const AudioSource * as, + double outQuality, + unsigned int outSampleRate = 0, + unsigned int outChannel = 0 ) + throw ( Exception ) + + : AudioEncoder ( as, + outQuality, + outSampleRate, + outChannel ) + { + init( sink); + } + /** * Copy constructor. * @@ -442,6 +493,11 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter $Source$ $Log$ + Revision 1.4 2002/03/28 16:47:38 darkeye + moved functions conv8() and conv16() to class Util (as conv()) + added resampling functionality + added support for variable bitrates + Revision 1.3 2001/10/19 12:39:42 darkeye created configure options to compile with or without lame / Ogg Vorbis