moved functions conv8() and conv16() to class Util (as conv())
added resampling functionality added support for variable bitrates
This commit is contained in:
darkeye
parent
a50598add0
commit
38dc041a4b
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@ -72,13 +72,38 @@ VorbisLibEncoder :: open ( void )
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}
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vorbis_info_init( &vorbisInfo);
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if ( (ret = vorbis_encode_init( &vorbisInfo,
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getInChannel(),
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getInSampleRate(),
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getOutBitrate() * 1000,
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getOutBitrate() * 1000,
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-1 )) ) {
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throw Exception( __FILE__, __LINE__, "vorbis encode init error", ret);
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if ( isVBR() ) {
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if ( (ret = vorbis_encode_init_vbr( &vorbisInfo,
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getInChannel(),
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getOutSampleRate(),
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getOutQuality() )) ) {
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throw Exception( __FILE__, __LINE__,
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"vorbis encode init error", ret);
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}
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} else {
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#ifdef VORBIS_LIB_RC3
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if ( (ret = vorbis_encode_init( &vorbisInfo,
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getInChannel(),
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getOutSampleRate(),
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getOutBitrate() * 1024,
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getOutBitrate() * 1024,
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-1 )) ) {
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throw Exception( __FILE__, __LINE__,
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"vorbis encode init error", ret);
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}
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#else
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if ( (ret = vorbis_encode_init( &vorbisInfo,
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getInChannel(),
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getOutSampleRate(),
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-1,
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getOutBitrate() * 1024,
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-1 )) ) {
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throw Exception( __FILE__, __LINE__,
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"vorbis encode init error", ret);
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}
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#endif
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}
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if ( (ret = vorbis_analysis_init( &vorbisDspState, &vorbisInfo)) ) {
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@ -100,8 +125,6 @@ VorbisLibEncoder :: open ( void )
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}
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// create an empty vorbis_comment structure
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vorbis_comment vorbisComment;
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vorbis_comment_init( &vorbisComment);
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// create the vorbis stream headers and send them to the underlying sink
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@ -117,8 +140,6 @@ VorbisLibEncoder :: open ( void )
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throw Exception( __FILE__, __LINE__, "vorbis header init error", ret);
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}
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vorbis_comment_init( &vorbisComment);
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ogg_stream_packetin( &oggStreamState, &header);
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ogg_stream_packetin( &oggStreamState, &commentHeader);
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ogg_stream_packetin( &oggStreamState, &codeHeader);
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@ -131,116 +152,16 @@ VorbisLibEncoder :: open ( void )
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vorbis_comment_clear( &vorbisComment );
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// initialize the resampling coverter if needed
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if ( converter ) {
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converter->initialize( resampleRatio, getInChannel());
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}
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encoderOpen = true;
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return true;
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}
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/*------------------------------------------------------------------------------
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* Convert an unsigned char buffer holding 8 bit PCM values with channels
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* interleaved to two float buffers (one for each channel)
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*----------------------------------------------------------------------------*/
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void
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VorbisLibEncoder :: conv8 ( unsigned char * pcmBuffer,
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unsigned int lenPcmBuffer,
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float * leftBuffer,
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float * rightBuffer,
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unsigned int channels )
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{
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if ( channels == 1 ) {
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unsigned int i, j;
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for ( i = 0, j = 0; i < lenPcmBuffer; ) {
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short int value;
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value = pcmBuffer[i++];
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leftBuffer[j] = ((float) value) / 128.f;
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++j;
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}
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} else {
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unsigned int i, j;
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for ( i = 0, j = 0; i < lenPcmBuffer; ) {
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short int value;
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value = pcmBuffer[i++];
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leftBuffer[j] = ((float) value) / 128.f;
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value = pcmBuffer[i++];
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rightBuffer[j] = ((float) value) / 128.f;
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++j;
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}
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}
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}
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/*------------------------------------------------------------------------------
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* Convert an unsigned char buffer holding 16 bit PCM values with channels
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* interleaved to two float buffers (one for each channel)
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*----------------------------------------------------------------------------*/
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void
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VorbisLibEncoder :: conv16 ( unsigned char * pcmBuffer,
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unsigned int lenPcmBuffer,
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float * leftBuffer,
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float * rightBuffer,
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unsigned int channels )
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{
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if ( isInBigEndian() ) {
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if ( channels == 1 ) {
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unsigned int i, j;
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for ( i = 0, j = 0; i < lenPcmBuffer; ) {
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short int value;
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value = pcmBuffer[i++] << 8;
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value |= pcmBuffer[i++];
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leftBuffer[j] = ((float) value) / 32768.f;
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++j;
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}
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} else {
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unsigned int i, j;
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for ( i = 0, j = 0; i < lenPcmBuffer; ) {
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short int value;
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value = pcmBuffer[i++] << 8;
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value |= pcmBuffer[i++];
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leftBuffer[j] = ((float) value) / 32768.f;
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value = pcmBuffer[i++] << 8;
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value |= pcmBuffer[i++];
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rightBuffer[j] = ((float) value) / 32768.f;
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++j;
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}
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}
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} else {
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if ( channels == 1 ) {
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unsigned int i, j;
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for ( i = 0, j = 0; i < lenPcmBuffer; ) {
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short int value;
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value = pcmBuffer[i++];
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value |= pcmBuffer[i++] << 8;
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leftBuffer[j] = ((float) value) / 32768.f;
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++j;
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}
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} else {
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unsigned int i, j;
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for ( i = 0, j = 0; i < lenPcmBuffer; ) {
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short int value;
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value = pcmBuffer[i++];
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value |= pcmBuffer[i++] << 8;
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leftBuffer[j] = ((float) value) / 32768.f;
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value = pcmBuffer[i++];
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value |= pcmBuffer[i++] << 8;
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rightBuffer[j] = ((float) value) / 32768.f;
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++j;
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}
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}
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}
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}
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/*------------------------------------------------------------------------------
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* Write data to the encoder
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@ -268,19 +189,37 @@ VorbisLibEncoder :: write ( const void * buf,
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unsigned int nSamples = processed / sampleSize;
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float ** vorbisBuffer;
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vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState, nSamples);
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if ( bitsPerSample == 8 ) {
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conv8( b, processed, vorbisBuffer[0], vorbisBuffer[1], channels);
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} else if ( bitsPerSample == 16 ) {
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conv16( b, processed, vorbisBuffer[0], vorbisBuffer[1], channels);
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// convert the byte-based raw input into a short buffer
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// with channels still interleaved
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unsigned int totalSamples = nSamples * channels;
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short int shortBuffer[totalSamples];
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Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
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if ( converter ) {
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// resample if needed
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int inCount = totalSamples;
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int outCount = (int) (inCount * resampleRatio);
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short int resampledBuffer[outCount * channels];
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int converted;
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converted = converter->resample( inCount,
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outCount,
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shortBuffer,
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resampledBuffer );
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vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState,
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converted / channels);
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Util::conv( resampledBuffer, converted, vorbisBuffer, channels);
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vorbis_analysis_wrote( &vorbisDspState, converted / channels);
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} else {
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throw Exception( __FILE__, __LINE__,
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"unsupported number of bits per sample for the encoder",
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bitsPerSample );
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vorbisBuffer = vorbis_analysis_buffer( &vorbisDspState, nSamples);
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Util::conv( shortBuffer, totalSamples, vorbisBuffer, channels);
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vorbis_analysis_wrote( &vorbisDspState, nSamples);
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}
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vorbis_analysis_wrote( &vorbisDspState, nSamples);
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vorbisBlocksOut();
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return processed;
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@ -315,9 +254,11 @@ VorbisLibEncoder :: vorbisBlocksOut ( void ) throw ()
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ogg_page oggPage;
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vorbis_analysis( &vorbisBlock, &oggPacket);
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#ifdef VORBIS_LIB_RC3
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vorbis_bitrate_addblock( &vorbisBlock);
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while ( vorbis_bitrate_flushpacket( &vorbisDspState, &oggPacket) ) {
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#endif
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ogg_stream_packetin( &oggStreamState, &oggPacket);
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@ -334,7 +275,9 @@ VorbisLibEncoder :: vorbisBlocksOut ( void ) throw ()
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oggPage.header_len + oggPage.body_len - written);
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}
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}
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#ifdef VORBIS_LIB_RC3
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}
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#endif
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}
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}
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@ -351,6 +294,7 @@ VorbisLibEncoder :: close ( void ) throw ( Exception )
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ogg_stream_clear( &oggStreamState);
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vorbis_block_clear( &vorbisBlock);
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vorbis_dsp_clear( &vorbisDspState);
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vorbis_comment_clear( &vorbisComment);
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vorbis_info_clear( &vorbisInfo);
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encoderOpen = false;
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@ -366,6 +310,11 @@ VorbisLibEncoder :: close ( void ) throw ( Exception )
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$Source$
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$Log$
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Revision 1.7 2002/03/28 16:47:38 darkeye
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moved functions conv8() and conv16() to class Util (as conv())
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added resampling functionality
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added support for variable bitrates
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Revision 1.6 2002/02/20 10:35:35 darkeye
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updated to work with Ogg Vorbis libs rc3 and current IceCast2 cvs
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@ -52,6 +52,7 @@
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#include "Reporter.h"
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#include "AudioEncoder.h"
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#include "Sink.h"
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#include "aflibConverter.h"
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/* ================================================================ constants */
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@ -93,6 +94,11 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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*/
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vorbis_block vorbisBlock;
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/**
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* Ogg Vorbis library global comment
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*/
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vorbis_comment vorbisComment;
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/**
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* Ogg library global stream state
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*/
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@ -103,6 +109,16 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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*/
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Ref<Sink> sink;
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/**
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* Resample ratio
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*/
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double resampleRatio;
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/**
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* aflibConverter object for possible resampling
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*/
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aflibConverter * converter;
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/**
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* Initialize the object.
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*
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@ -120,9 +136,17 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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getInBitsPerSample() );
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}
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if ( getOutSampleRate() != getInSampleRate() ) {
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throw Exception( __FILE__, __LINE__,
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"different in and out sample rate not supported");
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if ( getOutSampleRate() == getInSampleRate() ) {
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resampleRatio = 1;
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converter = 0;
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} else {
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resampleRatio = ( (double) getOutSampleRate() /
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(double) getInSampleRate() );
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// open the aflibConverter in
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// - high quality
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// - not linear (quadratic) interpolation
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// - not filter interpolation
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converter = new aflibConverter( true, true, false);
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}
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if ( getInChannel() != getOutChannel() ) {
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@ -143,42 +167,6 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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{
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}
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/**
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* Convert a char buffer holding 8 bit PCM values to a short buffer
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*
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* @param pcmBuffer buffer holding 8 bit PCM audio values,
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* channels are interleaved
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* @param lenPcmBuffer length of pcmBuffer
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* @param leftBuffer put the left channel here (must be big enough)
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* @param rightBuffer put the right channel here (if mono, not
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* touched, must be big enough)
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* @param channels number of channels (1 = mono, 2 = stereo)
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*/
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void
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conv8 ( unsigned char * pcmBuffer,
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unsigned int lenPcmBuffer,
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float * leftBuffer,
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float * rightBuffer,
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unsigned int channels );
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/**
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* Convert a char buffer holding 16 bit PCM values to a short buffer
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*
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* @param pcmBuffer buffer holding 16 bit PCM audio values,
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* channels are interleaved
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* @param lenPcmBuffer length of pcmBuffer
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* @param leftBuffer put the left channel here (must be big enough)
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* @param rightBuffer put the right channel here (if mono, not
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* touched, must be big enough)
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* @param channels number of channels (1 = mono, 2 = stereo)
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*/
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void
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conv16 ( unsigned char * pcmBuffer,
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unsigned int lenPcmBuffer,
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float * leftBuffer,
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float * rightBuffer,
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unsigned int channels );
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/**
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* Send pending Vorbis blocks to the underlying stream
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*/
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@ -203,13 +191,13 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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public:
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/**
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* Constructor.
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* Constructor for fixed bitrate encoding.
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*
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* @param sink the sink to send mp3 output to
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* @param inSampleRate sample rate of the input.
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* @param inBitsPerSample number of bits per sample of the input.
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* @param inChannel number of channels of the input.
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* @param outBitrate bit rate of the output (bits/sec).
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* @param outBitrate bit rate of the output (kbits/sec).
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* @param outSampleRate sample rate of the output.
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* If 0, inSampleRate is used.
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* @param outChannel number of channels of the output.
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@ -237,12 +225,46 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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}
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/**
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* Constructor.
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* Constructor for variable bitrate encoding.
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*
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* @param sink the sink to send mp3 output to
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* @param inSampleRate sample rate of the input.
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* @param inBitsPerSample number of bits per sample of the input.
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* @param inChannel number of channels of the input.
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* @param outQuality the quality of the stream (0.0 .. 1.0).
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* @param outSampleRate sample rate of the output.
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* If 0, inSampleRate is used.
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* @param outChannel number of channels of the output.
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* If 0, inChannel is used.
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* @exception Exception
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*/
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inline
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VorbisLibEncoder ( Sink * sink,
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unsigned int inSampleRate,
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unsigned int inBitsPerSample,
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unsigned int inChannel,
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double outQuality,
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unsigned int outSampleRate = 0,
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unsigned int outChannel = 0 )
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throw ( Exception )
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: AudioEncoder ( inSampleRate,
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inBitsPerSample,
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inChannel,
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outQuality,
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outSampleRate,
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outChannel )
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{
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init( sink);
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}
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/**
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* Constructor for fixed bitrate encoding.
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*
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* @param sink the sink to send mp3 output to
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* @param as get input sample rate, bits per sample and channels
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* from this AudioSource.
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* @param outBitrate bit rate of the output (bits/sec).
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* @param outBitrate bit rate of the output (kbits/sec).
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* @param outSampleRate sample rate of the output.
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* If 0, input sample rate is used.
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* @param outChannel number of channels of the output.
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@ -265,6 +287,35 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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init( sink);
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}
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/**
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* Constructor for variable bitrate encoding.
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*
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* @param sink the sink to send mp3 output to
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* @param as get input sample rate, bits per sample and channels
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* from this AudioSource.
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* @param outQuality the quality of the stream (0.0 .. 1.0).
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* @param outSampleRate sample rate of the output.
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* If 0, input sample rate is used.
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* @param outChannel number of channels of the output.
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* If 0, input channel is used.
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* @exception Exception
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*/
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inline
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VorbisLibEncoder ( Sink * sink,
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const AudioSource * as,
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double outQuality,
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unsigned int outSampleRate = 0,
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unsigned int outChannel = 0 )
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throw ( Exception )
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: AudioEncoder ( as,
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outQuality,
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outSampleRate,
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outChannel )
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{
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init( sink);
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}
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/**
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* Copy constructor.
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*
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@ -442,6 +493,11 @@ class VorbisLibEncoder : public AudioEncoder, public virtual Reporter
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$Source$
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$Log$
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||||
Revision 1.4 2002/03/28 16:47:38 darkeye
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moved functions conv8() and conv16() to class Util (as conv())
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added resampling functionality
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added support for variable bitrates
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Revision 1.3 2001/10/19 12:39:42 darkeye
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created configure options to compile with or without lame / Ogg Vorbis
|
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