make darkice compatile with the libaacplus 2.0.0 api
This commit is contained in:
rafael@riseup.net
parent
df521d34ba
commit
99f6ff8ad7
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@ -166,7 +166,7 @@ AC_ARG_WITH(aacplus-prefix,
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if test "x${USE_AACPLUS}" = "xyes" ; then
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AC_MSG_CHECKING( [for aacplus library at ${CONFIG_AACPLUS_PREFIX}] )
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LA_SEARCH_LIB( AACPLUS_LIB_LOC, AACPLUS_INC_LOC, libaacplus.a libaacplus.so, sbr_main.h,
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LA_SEARCH_LIB( AACPLUS_LIB_LOC, AACPLUS_INC_LOC, libaacplus.a libaacplus.so, aacplus.h,
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${CONFIG_AACPLUS_PREFIX})
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if test "x${AACPLUS_LIB_LOC}" != "x" ; then
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AC_DEFINE( HAVE_AACPLUS_LIB, 1, [build with aacplus library] )
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@ -76,91 +76,36 @@ aacPlusEncoder :: open ( void )
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"aacplus lib opening underlying sink error");
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}
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reportEvent(1, "Using aacplus codec version", "720 3gpp");
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reportEvent(1, "Using aacplus codec");
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bitrate = getOutBitrate() * 1000;
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bandwidth = 0;
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useParametricStereo = 0;
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numAncDataBytes=0;
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coreWriteOffset = 0;
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envReadOffset = 0;
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writeOffset = INPUT_DELAY*MAX_CHANNELS;
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writtenSamples = 0;
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aacEnc = NULL;
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hEnvEnc=NULL;
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encoderHandle = aacplusEncOpen(getOutSampleRate(),
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getInChannel(),
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&inputSamples,
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&maxOutputBytes);
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/* set up basic parameters for aacPlus codec */
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AacInitDefaultConfig(&config);
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nChannelsAAC = nChannelsSBR = getOutChannel();
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aacplusEncConfiguration * aacplusConfig;
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if ( (getInChannel() == 2) && (bitrate >= 16000) && (bitrate < 44001) ) {
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useParametricStereo = 1;
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nChannelsAAC = 1;
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nChannelsSBR = 2;
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aacplusConfig = aacplusEncGetCurrentConfiguration(encoderHandle);
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reportEvent(10, "use Parametric Stereo");
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aacplusConfig->bitRate = getOutBitrate() * 1000;
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aacplusConfig->bandWidth = lowpass;
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aacplusConfig->outputFormat = 1;
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aacplusConfig->inputFormat = AACPLUS_INPUT_16BIT;
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aacplusConfig->nChannelsOut = getOutChannel();
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envReadOffset = (MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS;
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coreWriteOffset = CORE_INPUT_OFFSET_PS;
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writeOffset = envReadOffset;
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} else {
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/* set up 2:1 downsampling */
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InitIIR21_Resampler(&(IIR21_reSampler[0]));
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InitIIR21_Resampler(&(IIR21_reSampler[1]));
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if(IIR21_reSampler[0].delay > MAX_DS_FILTER_DELAY)
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throw Exception(__FILE__, __LINE__, "IIR21 resampler delay is bigger then MAX_DS_FILTER_DELAY");
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writeOffset += IIR21_reSampler[0].delay*MAX_CHANNELS;
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if (!aacplusEncSetConfiguration(encoderHandle, aacplusConfig)) {
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throw Exception(__FILE__, __LINE__,
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"error configuring libaacplus library");
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}
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sampleRateAAC = getOutSampleRate();
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config.bitRate = bitrate;
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config.nChannelsIn=getInChannel();
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config.nChannelsOut=nChannelsAAC;
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config.bandWidth=bandwidth;
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/* set up SBR configuration */
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if(!IsSbrSettingAvail(bitrate, nChannelsAAC, sampleRateAAC, &sampleRateAAC))
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throw Exception(__FILE__, __LINE__, "No valid SBR configuration found");
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InitializeSbrDefaults (&sbrConfig);
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sbrConfig.usePs = useParametricStereo;
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AdjustSbrSettings( &sbrConfig,
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bitrate,
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nChannelsAAC,
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sampleRateAAC,
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AACENC_TRANS_FAC,
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24000);
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EnvOpen( &hEnvEnc,
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inBuf + coreWriteOffset,
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&sbrConfig,
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&config.bandWidth);
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/* set up AAC encoder, now that samling rate is known */
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config.sampleRate = sampleRateAAC;
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if (AacEncOpen(&aacEnc, config) != 0){
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AacEncClose(aacEnc);
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throw Exception(__FILE__, __LINE__, "Initialisation of AAC failed !");
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}
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init_plans();
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/* create the ADTS header */
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adts_hdr(outBuf, &config);
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inSamples = AACENC_BLOCKSIZE * getInChannel() * 2;
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// initialize the resampling coverter if needed
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if ( converter ) {
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#ifdef HAVE_SRC_LIB
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converterData.input_frames = 4096/((getInBitsPerSample() / 8) * getInChannel());
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converterData.data_in = new float[converterData.input_frames*getInChannel()];
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converterData.output_frames = (int) (converterData.input_frames * resampleRatio + 1);
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if ((int) inSamples > getInChannel() * converterData.output_frames) {
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resampledOffset = new float[2 * inSamples];
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if ((int) inputSamples > getInChannel() * converterData.output_frames) {
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resampledOffset = new float[2 * inputSamples];
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} else {
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resampledOffset = new float[2 * getInChannel() * converterData.input_frames];
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}
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@ -178,13 +123,9 @@ aacPlusEncoder :: open ( void )
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}
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aacplusOpen = true;
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reportEvent(10, "bitrate=", bitrate);
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reportEvent(10, "nChannelsIn", getInChannel());
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reportEvent(10, "nChannelsOut", getOutChannel());
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reportEvent(10, "nChannelsSBR", nChannelsSBR);
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reportEvent(10, "nChannelsAAC", nChannelsAAC);
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reportEvent(10, "sampleRateAAC", sampleRateAAC);
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reportEvent(10, "inSamples", inSamples);
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reportEvent(10, "nChannelsAAC", aacplusConfig->nChannelsOut);
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reportEvent(10, "sampleRateAAC", aacplusConfig->sampleRate);
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reportEvent(10, "inSamples", inputSamples);
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return true;
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}
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@ -203,17 +144,19 @@ aacPlusEncoder :: write ( const void * buf,
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unsigned int channels = getInChannel();
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unsigned int bitsPerSample = getInBitsPerSample();
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unsigned int sampleSize = (bitsPerSample / 8) * channels;
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unsigned char * b = (unsigned char*) buf;
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unsigned int processed = len - (len % sampleSize);
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unsigned int nSamples = processed / sampleSize;
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unsigned int samples = (unsigned int) nSamples * channels;
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int processedSamples = 0;
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unsigned char * aacplusBuf = new unsigned char[maxOutputBytes];
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int samples = (int) nSamples * channels;
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int processedSamples = 0;
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if ( converter ) {
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unsigned int converted;
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#ifdef HAVE_SRC_LIB
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src_short_to_float_array ((short *) buf, converterData.data_in, samples);
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src_short_to_float_array ((short *) b, converterData.data_in, samples);
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converterData.input_frames = nSamples;
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converterData.data_out = resampledOffset + (resampledOffsetSize * channels);
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int srcError = src_process (converter, &converterData);
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@ -224,7 +167,6 @@ aacPlusEncoder :: write ( const void * buf,
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int inCount = nSamples;
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short int * shortBuffer = new short int[samples];
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int outCount = (int) (inCount * resampleRatio);
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unsigned char * b = (unsigned char*) buf;
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Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
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converted = converter->resample( inCount,
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outCount+1,
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@ -235,18 +177,27 @@ aacPlusEncoder :: write ( const void * buf,
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resampledOffsetSize += converted;
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// encode samples (if enough)
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while(resampledOffsetSize - processedSamples >= inSamples/channels) {
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while(resampledOffsetSize - processedSamples >= inputSamples/channels) {
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int outputBytes;
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#ifdef HAVE_SRC_LIB
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short *shortData = new short[inSamples];
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short *shortData = new short[inputSamples];
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src_float_to_short_array(resampledOffset + (processedSamples * channels),
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shortData, inSamples) ;
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encodeAacSamples (shortData, inSamples, channels);
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shortData, inputSamples) ;
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outputBytes = aacplusEncEncode(encoderHandle,
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(int32_t*) shortData,
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inputSamples,
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aacplusBuf,
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maxOutputBytes);
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delete [] shortData;
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#else
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encodeAacSamples (&resampledOffset[processedSamples*channels], inSamples, channels);
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outputBytes = aacplusEncEncode(encoderHandle,
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(int32_t*) &resampledOffset[processedSamples*channels],
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inputSamples,
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aacplusBuf,
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maxOutputBytes);
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#endif
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processedSamples+=inSamples/channels;
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getSink()->write(aacplusBuf, outputBytes);
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processedSamples+=inputSamples/channels;
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}
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if (processedSamples && (int) resampledOffsetSize >= processedSamples) {
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@ -262,72 +213,29 @@ aacPlusEncoder :: write ( const void * buf,
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#endif
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}
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} else {
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encodeAacSamples ((short *) buf, samples, channels);
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while (processedSamples < samples) {
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int outputBytes;
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int inSamples = samples - processedSamples < (int) inputSamples
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? samples - processedSamples
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: inputSamples;
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outputBytes = aacplusEncEncode(encoderHandle,
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(int32_t*) (b + processedSamples/sampleSize),
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inSamples,
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aacplusBuf,
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maxOutputBytes);
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getSink()->write(aacplusBuf, outputBytes);
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processedSamples += inSamples;
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}
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}
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delete[] aacplusBuf;
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// return processedSamples;
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return samples;
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}
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void
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aacPlusEncoder :: encodeAacSamples (short *TimeDataPcm, unsigned int samples, int channels)
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throw ( Exception )
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{
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unsigned int i;
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int ch, outSamples, numOutBytes;
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for (i=0; i<samples; i++)
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inBuf[(2/channels)*i+writeOffset+writtenSamples] = (float) TimeDataPcm[i];
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writtenSamples+=samples;
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if (writtenSamples < inSamples)
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return;
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/* encode one SBR frame */
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EnvEncodeFrame( hEnvEnc,
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inBuf + envReadOffset,
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inBuf + coreWriteOffset,
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MAX_CHANNELS,
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&numAncDataBytes,
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ancDataBytes);
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/* 2:1 downsampling for AAC core */
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if (!useParametricStereo) {
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for( ch=0; ch<nChannelsAAC; ch++ )
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IIR21_Downsample( &(IIR21_reSampler[ch]),
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inBuf + writeOffset+ch,
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writtenSamples/channels,
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MAX_CHANNELS,
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inBuf+ch,
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&outSamples,
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MAX_CHANNELS);
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}
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/* encode one AAC frame */
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AacEncEncode( aacEnc,
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inBuf,
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useParametricStereo ? 1 : MAX_CHANNELS, /* stride (step) */
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ancDataBytes,
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&numAncDataBytes,
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(unsigned *) (outBuf+ADTS_HEADER_SIZE),
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&numOutBytes);
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if (useParametricStereo) {
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memcpy( inBuf,inBuf+AACENC_BLOCKSIZE,CORE_INPUT_OFFSET_PS*sizeof(float));
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} else {
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memmove( inBuf,inBuf+AACENC_BLOCKSIZE*2*MAX_CHANNELS,writeOffset*sizeof(float));
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}
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/* Write one frame of encoded audio */
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if (numOutBytes) {
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adts_hdr_up(outBuf, numOutBytes);
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sink->write(outBuf, numOutBytes+ADTS_HEADER_SIZE);
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}
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writtenSamples=0;
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return;
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}
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/*------------------------------------------------------------------------------
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* Flush the data from the encoder
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*----------------------------------------------------------------------------*/
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@ -352,12 +260,7 @@ aacPlusEncoder :: close ( void ) throw ( Exception )
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if ( isOpen() ) {
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flush();
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destroy_plans();
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AacEncClose(aacEnc);
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if (hEnvEnc) {
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EnvClose(hEnvEnc);
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}
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aacplusEncClose(encoderHandle);
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aacplusOpen = false;
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sink->close();
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@ -41,18 +41,7 @@
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#endif
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#ifdef HAVE_AACPLUS_LIB
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extern "C" {
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#include <libaacplus/cfftn.h>
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#include <libaacplus/FloatFR.h>
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#include <libaacplus/aacenc.h>
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#include <libaacplus/resampler.h>
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#include <libaacplus/adts.h>
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#include <libaacplus/sbr_main.h>
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#include <libaacplus/aac_ram.h>
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#include <libaacplus/aac_rom.h>
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}
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#include <aacplus.h>
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#else
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#error configure with aacplus
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#endif
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@ -87,12 +76,6 @@ extern "C" {
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* @version $Revision$
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*/
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#define CORE_DELAY (1600)
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#define INPUT_DELAY ((CORE_DELAY)*2 +6*64-2048+1) /* ((1600 (core codec)*2 (multi rate) + 6*64 (sbr dec delay) - 2048 (sbr enc delay) + magic*/
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#define MAX_DS_FILTER_DELAY 16 /* the additional max resampler filter delay (source fs)*/
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#define CORE_INPUT_OFFSET_PS (0) /* (96-64) makes AAC still some 64 core samples too early wrt SBR ... maybe -32 would be even more correct, but 1024-32 would need additional SBR bitstream delay by one frame */
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class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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{
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private:
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@ -124,31 +107,26 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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*/
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Ref<Sink> sink;
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float inBuf[(AACENC_BLOCKSIZE*2 + MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS];
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char outBuf[(6144/8)*MAX_CHANNELS+ADTS_HEADER_SIZE];
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IIR21_RESAMPLER IIR21_reSampler[MAX_CHANNELS];
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/**
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* The handle to the AAC+ encoder instance.
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*/
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aacplusEncHandle encoderHandle;
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AACENC_CONFIG config;
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/**
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* The maximum number of input samples to supply to the encoder.
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*/
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unsigned long inputSamples;
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int nChannelsAAC, nChannelsSBR;
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unsigned int sampleRateAAC;
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/**
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* The maximum number of output bytes the encoder returns in one call.
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*/
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unsigned long maxOutputBytes;
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int bitrate;
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int bandwidth;
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unsigned int numAncDataBytes;
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unsigned char ancDataBytes[MAX_PAYLOAD_SIZE];
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bool useParametricStereo;
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int coreWriteOffset;
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int envReadOffset;
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int writeOffset;
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struct AAC_ENCODER *aacEnc;
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unsigned int inSamples;
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unsigned int writtenSamples;
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HANDLE_SBR_ENCODER hEnvEnc;
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sbrConfiguration sbrConfig;
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/**
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* Lowpass filter. Sound frequency in Hz, from where up the
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* input is cut.
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*/
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int lowpass;
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/**
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* Initialize the object.
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@ -157,10 +135,11 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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* @exception Exception
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*/
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inline void
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init ( Sink * sink) throw (Exception)
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init ( Sink * sink, int lowpass) throw (Exception)
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{
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this->aacplusOpen = false;
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this->sink = sink;
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this->lowpass = lowpass;
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/* TODO: if we have float as input, we don't need conversion */
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if ( getInBitsPerSample() != 16 && getInBitsPerSample() != 32 ) {
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@ -179,11 +158,6 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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"unsupported number of output channels for the encoder",
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getOutChannel() );
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}
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/* TODO: this will be neede when we implement mono aac+ encoding */
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if ( getInChannel() != getOutChannel() ) {
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throw Exception( __FILE__, __LINE__,
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"input channels and output channels do not match");
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}
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if ( getOutSampleRate() == getInSampleRate() ) {
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resampleRatio = 1;
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@ -237,17 +211,6 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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"specified bits per sample with samplerate conversion not supported",
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getInBitsPerSample() );
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}
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bitrate = getOutBitrate() * 1000;
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bandwidth = 0;
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useParametricStereo = 0;
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numAncDataBytes=0;
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coreWriteOffset = 0;
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envReadOffset = 0;
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writeOffset = INPUT_DELAY*MAX_CHANNELS;
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writtenSamples = 0;
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aacEnc = NULL;
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hEnvEnc=NULL;
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}
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/**
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@ -269,10 +232,6 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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}
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}
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void
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encodeAacSamples (short *TimeDataPcm, unsigned int samples, int channels)
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throw ( Exception );
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protected:
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/**
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@ -335,7 +294,7 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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outSampleRate,
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outChannel )
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{
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init( sink);
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init( sink, lowpass);
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}
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/**
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@ -376,7 +335,7 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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outSampleRate,
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outChannel )
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{
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init( sink);
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init( sink, lowpass );
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}
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/**
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@ -389,7 +348,7 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
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throw ( Exception )
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: AudioEncoder( encoder )
|
||||
{
|
||||
init( encoder.sink.get());
|
||||
init( encoder.sink.get(), encoder.lowpass);
|
||||
}
|
||||
|
||||
|
||||
|
|
@ -420,7 +379,7 @@ class aacPlusEncoder : public AudioEncoder, public virtual Reporter
|
|||
if ( this != &encoder ) {
|
||||
strip();
|
||||
AudioEncoder::operator=( encoder);
|
||||
init( encoder.sink.get());
|
||||
init( encoder.sink.get(), encoder.lowpass);
|
||||
}
|
||||
|
||||
return *this;
|
||||
|
|
|
|||
Loading…
Reference in New Issue