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added svn keywords, tabs changed into 4 whitespaces, re #2

This commit is contained in:
tipok 2008-11-05 13:02:24 +00:00
parent c8260ff827
commit d606610431
2 changed files with 96 additions and 95 deletions
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187
darkice/branches/darkice-aacp/src/aacPlusEncoder.cpp
View File

@ -77,31 +77,30 @@ aacPlusEncoder :: open ( void )
} }
reportEvent(1, "Using aacplus codec version", "720 3gpp"); reportEvent(1, "Using aacplus codec version", "720 3gpp");
/* set up basic parameters for aacPlus codec */ /* set up basic parameters for aacPlus codec */
AacInitDefaultConfig(&config); AacInitDefaultConfig(&config);
nChannelsAAC = nChannelsSBR = getOutChannel(); nChannelsAAC = nChannelsSBR = getOutChannel();
if ( (getInChannel() == 2) && (bitrate >= 16000) && (bitrate < 44001) ) { if ( (getInChannel() == 2) && (bitrate >= 16000) && (bitrate < 44001) ) {
useParametricStereo = 1; useParametricStereo = 1;
nChannelsAAC = 1; nChannelsAAC = 1;
nChannelsSBR = 2; nChannelsSBR = 2;
reportEvent(10, "use Parametric Stereo"); reportEvent(10, "use Parametric Stereo");
envReadOffset = (MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS; envReadOffset = (MAX_DS_FILTER_DELAY + INPUT_DELAY)*MAX_CHANNELS;
coreWriteOffset = CORE_INPUT_OFFSET_PS; coreWriteOffset = CORE_INPUT_OFFSET_PS;
writeOffset = envReadOffset; writeOffset = envReadOffset;
} else { } else {
/* set up 2:1 downsampling */ /* set up 2:1 downsampling */
InitIIR21_Resampler(&(IIR21_reSampler[0])); InitIIR21_Resampler(&(IIR21_reSampler[0]));
InitIIR21_Resampler(&(IIR21_reSampler[1])); InitIIR21_Resampler(&(IIR21_reSampler[1]));
if(IIR21_reSampler[0].delay > MAX_DS_FILTER_DELAY){ if(IIR21_reSampler[0].delay > MAX_DS_FILTER_DELAY)
throw Exception(__FILE__, __LINE__, "IIR21 resampler delay is bigger then MAX_DS_FILTER_DELAY"); throw Exception(__FILE__, __LINE__, "IIR21 resampler delay is bigger then MAX_DS_FILTER_DELAY");
} writeOffset += IIR21_reSampler[0].delay*MAX_CHANNELS;
writeOffset += IIR21_reSampler[0].delay*MAX_CHANNELS;
} }
sampleRateAAC = getInSampleRate(); sampleRateAAC = getInSampleRate();
@ -110,31 +109,30 @@ aacPlusEncoder :: open ( void )
config.nChannelsOut=nChannelsAAC; config.nChannelsOut=nChannelsAAC;
config.bandWidth=bandwidth; config.bandWidth=bandwidth;
/* set up SBR configuration */ /* set up SBR configuration */
if(!IsSbrSettingAvail (bitrate, nChannelsAAC, sampleRateAAC, &sampleRateAAC)) { if(!IsSbrSettingAvail(bitrate, nChannelsAAC, sampleRateAAC, &sampleRateAAC))
throw Exception(__FILE__, __LINE__, "No valid SBR configuration found"); throw Exception(__FILE__, __LINE__, "No valid SBR configuration found");
}
InitializeSbrDefaults (&sbrConfig); InitializeSbrDefaults (&sbrConfig);
sbrConfig.usePs = useParametricStereo; sbrConfig.usePs = useParametricStereo;
AdjustSbrSettings( &sbrConfig, AdjustSbrSettings( &sbrConfig,
bitrate, bitrate,
nChannelsAAC, nChannelsAAC,
sampleRateAAC, sampleRateAAC,
AACENC_TRANS_FAC, AACENC_TRANS_FAC,
24000); 24000);
EnvOpen( &hEnvEnc,
inBuf + coreWriteOffset,
&sbrConfig,
&config.bandWidth);
EnvOpen( &hEnvEnc,
inBuf + coreWriteOffset,
&sbrConfig,
&config.bandWidth);
/* set up AAC encoder, now that samling rate is known */ /* set up AAC encoder, now that samling rate is known */
config.sampleRate = sampleRateAAC; config.sampleRate = sampleRateAAC;
if (AacEncOpen(&aacEnc, config) != 0){ if (AacEncOpen(&aacEnc, config) != 0){
AacEncClose(aacEnc); AacEncClose(aacEnc);
throw Exception(__FILE__, __LINE__, "Initialisation of AAC failed !"); throw Exception(__FILE__, __LINE__, "Initialisation of AAC failed !");
} }
init_plans(); init_plans();
@ -182,73 +180,76 @@ aacPlusEncoder :: write ( const void * buf,
reportEvent(10, "converting short to float"); reportEvent(10, "converting short to float");
short *TimeDataPcm = (short *) buf; short *TimeDataPcm = (short *) buf;
for (i=0; i<samples; i++) for (i=0; i<samples; i++)
inBuf[i+writeOffset+writtenSamples] = (float) TimeDataPcm[i]; inBuf[i+writeOffset+writtenSamples] = (float) TimeDataPcm[i];
writtenSamples+=samples; writtenSamples+=samples;
reportEvent(10, "writtenSamples", writtenSamples); reportEvent(10, "writtenSamples", writtenSamples);
if (writtenSamples < inSamples) if (writtenSamples < inSamples)
return samples; return samples;
/* encode one SBR frame */ /* encode one SBR frame */
reportEvent(10, "encode one SBR frame"); reportEvent(10, "encode one SBR frame");
EnvEncodeFrame( hEnvEnc, EnvEncodeFrame( hEnvEnc,
inBuf + envReadOffset, inBuf + envReadOffset,
inBuf + coreWriteOffset, inBuf + coreWriteOffset,
MAX_CHANNELS, MAX_CHANNELS,
&numAncDataBytes, &numAncDataBytes,
ancDataBytes); ancDataBytes);
reportEvent(10, "numAncDataBytes=", numAncDataBytes); reportEvent(10, "numAncDataBytes=", numAncDataBytes);
/* 2:1 downsampling for AAC core */ /* 2:1 downsampling for AAC core */
if (!useParametricStereo){ if (!useParametricStereo) {
reportEvent(10, "2:1 downsampling for AAC core"); reportEvent(10, "2:1 downsampling for AAC core");
for( ch=0; ch<nChannelsAAC; ch++ ) for( ch=0; ch<nChannelsAAC; ch++ )
IIR21_Downsample( &(IIR21_reSampler[ch]), IIR21_Downsample( &(IIR21_reSampler[ch]),
inBuf+writeOffset+ch, inBuf + writeOffset+ch,
writtenSamples/getInChannel(), writtenSamples/getInChannel(),
MAX_CHANNELS, MAX_CHANNELS,
inBuf+ch, inBuf+ch,
&outSamples, &outSamples,
MAX_CHANNELS); MAX_CHANNELS);
reportEvent(10, "outSamples=", outSamples);
reportEvent(10, "outSamples=", outSamples);
} }
/* encode one AAC frame */ /* encode one AAC frame */
if (hEnvEnc && useParametricStereo) { if (hEnvEnc && useParametricStereo) {
reportEvent(10, "Parametric Stereo encode one AAC frame"); reportEvent(10, "Parametric Stereo encode one AAC frame");
AacEncEncode( aacEnc, AacEncEncode( aacEnc,
inBuf, inBuf,
1, /* stride (step) */ 1, /* stride (step) */
ancDataBytes, ancDataBytes,
&numAncDataBytes, &numAncDataBytes,
(unsigned *) (outBuf+ADTS_HEADER_SIZE), (unsigned *) (outBuf+ADTS_HEADER_SIZE),
&numOutBytes); &numOutBytes);
if(hEnvEnc)
if(hEnvEnc) memcpy( inBuf,inBuf+AACENC_BLOCKSIZE,CORE_INPUT_OFFSET_PS*sizeof(float));
memcpy( inBuf,inBuf+AACENC_BLOCKSIZE,CORE_INPUT_OFFSET_PS*sizeof(float));
} else { } else {
reportEvent(10, "encode one AAC frame"); reportEvent(10, "encode one AAC frame");
AacEncEncode( aacEnc, AacEncEncode( aacEnc,
inBuf+coreReadOffset, inBuf + coreReadOffset,
MAX_CHANNELS, MAX_CHANNELS,
ancDataBytes, ancDataBytes,
&numAncDataBytes, &numAncDataBytes,
(unsigned *) (outBuf+ADTS_HEADER_SIZE), (unsigned *) (outBuf+ADTS_HEADER_SIZE),
&numOutBytes); &numOutBytes);
reportEvent(10, "done AAC=", numOutBytes); reportEvent(10, "done AAC=", numOutBytes);
if(hEnvEnc) if(hEnvEnc)
memmove( inBuf,inBuf+AACENC_BLOCKSIZE*2*MAX_CHANNELS,writeOffset*sizeof(float)); memmove( inBuf,inBuf+AACENC_BLOCKSIZE*2*MAX_CHANNELS,writeOffset*sizeof(float));
} }
/* Write one frame of encoded audio */ /* Write one frame of encoded audio */
if (numOutBytes) { if (numOutBytes) {
reportEvent(10, "Write one frame of encoded audio:", numOutBytes+ADTS_HEADER_SIZE); reportEvent(10, "Write one frame of encoded audio:", numOutBytes+ADTS_HEADER_SIZE);
adts_hdr_up(outBuf, numOutBytes); adts_hdr_up(outBuf, numOutBytes);
sink->write(outBuf, numOutBytes+ADTS_HEADER_SIZE); sink->write(outBuf, numOutBytes+ADTS_HEADER_SIZE);
} }
writtenSamples=0; writtenSamples=0;
return samples; return samples;
@ -278,16 +279,16 @@ aacPlusEncoder :: close ( void ) throw ( Exception )
{ {
if ( isOpen() ) { if ( isOpen() ) {
flush(); flush();
destroy_plans(); destroy_plans();
AacEncClose(aacEnc); AacEncClose(aacEnc);
if (hEnvEnc) { if (hEnvEnc) {
EnvClose(hEnvEnc); EnvClose(hEnvEnc);
} }
aacplusOpen = false; aacplusOpen = false;
sink->close(); sink->close();
} }
} }

4
darkice/branches/darkice-aacp/src/aacPlusEncoder.h
View File

@ -78,8 +78,8 @@ extern "C" {
/** /**
* A class representing aacplus AAC+ encoder. * A class representing aacplus AAC+ encoder.
* *
* @author $Author: tipok $ * @author $Author$
* @version $Revision: 1 $ * @version $Revision$
*/ */
#define CORE_DELAY (1600) #define CORE_DELAY (1600)