WIP: support 64 samples frame size in the client
This commit is contained in:
parent
f6ea80ca81
commit
5f85660cde
4 changed files with 150 additions and 118 deletions
152
src/client.cpp
152
src/client.cpp
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@ -51,7 +51,7 @@ CClient::CClient ( const quint16 iPortNumber,
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CurOpusDecoder ( nullptr ),
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CurOpusDecoder ( nullptr ),
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eAudioCompressionType ( CT_OPUS ),
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eAudioCompressionType ( CT_OPUS ),
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iCeltNumCodedBytes ( OPUS_NUM_BYTES_MONO_LOW_QUALITY ),
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iCeltNumCodedBytes ( OPUS_NUM_BYTES_MONO_LOW_QUALITY ),
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iOPUSFrameSizeSamples ( SYSTEM_FRAME_SIZE_SAMPLES ),
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iOPUSFrameSizeSamples ( DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES ),
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eAudioQuality ( AQ_NORMAL ),
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eAudioQuality ( AQ_NORMAL ),
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eAudioChannelConf ( CC_MONO ),
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eAudioChannelConf ( CC_MONO ),
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iNumAudioChannels ( 1 ),
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iNumAudioChannels ( 1 ),
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@ -669,16 +669,16 @@ void CClient::Stop()
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void CClient::Init()
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void CClient::Init()
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{
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{
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// check if possible frame size factors are supported
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// check if possible frame size factors are supported
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const int iFraSizePreffered = FRAME_SIZE_FACTOR_PREFERRED * SYSTEM_FRAME_SIZE_SAMPLES;
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const int iFraSizePreffered = SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_PREFERRED;
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const int iFraSizeDefault = FRAME_SIZE_FACTOR_DEFAULT * SYSTEM_FRAME_SIZE_SAMPLES;
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const int iFraSizeDefault = SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_DEFAULT;
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const int iFraSizeSafe = FRAME_SIZE_FACTOR_SAFE * SYSTEM_FRAME_SIZE_SAMPLES;
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const int iFraSizeSafe = SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_SAFE;
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bFraSiFactPrefSupported = ( Sound.Init ( iFraSizePreffered ) == iFraSizePreffered );
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bFraSiFactPrefSupported = ( Sound.Init ( iFraSizePreffered ) == iFraSizePreffered );
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bFraSiFactDefSupported = ( Sound.Init ( iFraSizeDefault ) == iFraSizeDefault );
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bFraSiFactDefSupported = ( Sound.Init ( iFraSizeDefault ) == iFraSizeDefault );
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bFraSiFactSafeSupported = ( Sound.Init ( iFraSizeSafe ) == iFraSizeSafe );
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bFraSiFactSafeSupported = ( Sound.Init ( iFraSizeSafe ) == iFraSizeSafe );
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// translate block size index in actual block size
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// translate block size index in actual block size
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const int iPrefMonoFrameSize = iSndCrdPrefFrameSizeFactor * SYSTEM_FRAME_SIZE_SAMPLES;
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const int iPrefMonoFrameSize = iSndCrdPrefFrameSizeFactor * SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
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// get actual sound card buffer size using preferred size
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// get actual sound card buffer size using preferred size
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iMonoBlockSizeSam = Sound.Init ( iPrefMonoFrameSize );
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iMonoBlockSizeSam = Sound.Init ( iPrefMonoFrameSize );
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@ -686,12 +686,12 @@ void CClient::Init()
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// Calculate the current sound card frame size factor. In case
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// Calculate the current sound card frame size factor. In case
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// the current mono block size is not a multiple of the system
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// the current mono block size is not a multiple of the system
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// frame size, we have to use a sound card conversion buffer.
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// frame size, we have to use a sound card conversion buffer.
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if ( ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED ) ) ||
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if ( ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_PREFERRED ) ) ||
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( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT ) ) ||
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( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_DEFAULT ) ) ||
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( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE ) ) )
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( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_SAFE ) ) )
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{
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{
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// regular case: one of our predefined buffer sizes is available
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// regular case: one of our predefined buffer sizes is available
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iSndCrdFrameSizeFactor = iMonoBlockSizeSam / SYSTEM_FRAME_SIZE_SAMPLES;
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iSndCrdFrameSizeFactor = iMonoBlockSizeSam / SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
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// no sound card conversion buffer required
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// no sound card conversion buffer required
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bSndCrdConversionBufferRequired = false;
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bSndCrdConversionBufferRequired = false;
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@ -700,51 +700,49 @@ void CClient::Init()
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{
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{
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// An unsupported sound card buffer size is currently used -> we have
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// An unsupported sound card buffer size is currently used -> we have
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// to use a conversion buffer. Per definition we use the smallest buffer
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// to use a conversion buffer. Per definition we use the smallest buffer
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// size as the current frame size
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// size as the current frame size.
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// store actual sound card buffer size (stereo)
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// store actual sound card buffer size (stereo)
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iSndCardMonoBlockSizeSamConvBuff = iMonoBlockSizeSam;
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const int iSndCardStereoBlockSizeSamConvBuff = 2 * iMonoBlockSizeSam;
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// overwrite block size by smallest supported buffer size
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iSndCrdFrameSizeFactor = FRAME_SIZE_FACTOR_PREFERRED;
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iMonoBlockSizeSam = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED;
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iStereoBlockSizeSam = 2 * iMonoBlockSizeSam;
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// inits for conversion buffer (the size of the conversion buffer must
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// be the sum of input/output sizes which is the worst case fill level)
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const int iConBufSize = iStereoBlockSizeSam + iSndCardStereoBlockSizeSamConvBuff;
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SndCrdConversionBufferIn.Init ( iConBufSize );
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SndCrdConversionBufferOut.Init ( iConBufSize );
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vecDataConvBuf.Init ( iStereoBlockSizeSam );
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// the output conversion buffer must be filled with the inner
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// block size for initialization (this is the latency which is
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// introduced by the conversion buffer) to avoid buffer underruns
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const CVector<int16_t> vZeros ( iStereoBlockSizeSam, 0 );
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SndCrdConversionBufferOut.Put ( vZeros, vZeros.Size() );
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bSndCrdConversionBufferRequired = true;
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bSndCrdConversionBufferRequired = true;
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iSndCardMonoBlockSizeSamConvBuff = iMonoBlockSizeSam;
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// overwrite block size factor by using one frame
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iSndCrdFrameSizeFactor = 1;
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}
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}
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// calculate stereo (two channels) buffer size
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// select the OPUS frame size mode depending on current mono block size samples
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iStereoBlockSizeSam = 2 * iMonoBlockSizeSam;
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if ( bSndCrdConversionBufferRequired )
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{
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if ( iSndCardMonoBlockSizeSamConvBuff < DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES )
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{
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iMonoBlockSizeSam = SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
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eAudioCompressionType = CT_OPUS64;
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}
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else
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{
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iMonoBlockSizeSam = DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES;
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eAudioCompressionType = CT_OPUS;
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}
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}
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else
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{
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if ( iMonoBlockSizeSam < DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES )
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{
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eAudioCompressionType = CT_OPUS64;
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}
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else
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{
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// since we use double size frame size for OPUS, we have to adjust the frame size factor
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iSndCrdFrameSizeFactor /= 2;
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eAudioCompressionType = CT_OPUS;
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// init reverberation
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}
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AudioReverbL.Init ( SYSTEM_SAMPLE_RATE_HZ );
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}
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AudioReverbR.Init ( SYSTEM_SAMPLE_RATE_HZ );
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// inits for audio coding
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// inits for audio coding
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// always use the OPUS codec
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if ( eAudioCompressionType == CT_OPUS )
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#if ( SYSTEM_FRAME_SIZE_SAMPLES == 64 )
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{
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eAudioCompressionType = CT_OPUS64;
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iOPUSFrameSizeSamples = DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES;
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#else
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eAudioCompressionType = CT_OPUS;
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#endif
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iOPUSFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
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if ( eAudioChannelConf == CC_MONO )
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if ( eAudioChannelConf == CC_MONO )
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{
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{
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@ -752,6 +750,37 @@ void CClient::Init()
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CurOpusDecoder = OpusDecoderMono;
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CurOpusDecoder = OpusDecoderMono;
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iNumAudioChannels = 1;
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iNumAudioChannels = 1;
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switch ( eAudioQuality )
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{
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case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_LOW_QUALITY_DBLE_FRAMESIZE; break;
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case AQ_NORMAL: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_NORMAL_QUALITY_DBLE_FRAMESIZE; break;
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case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE; break;
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}
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}
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else
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{
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CurOpusEncoder = OpusEncoderStereo;
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CurOpusDecoder = OpusDecoderStereo;
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iNumAudioChannels = 2;
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switch ( eAudioQuality )
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{
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case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_LOW_QUALITY_DBLE_FRAMESIZE; break;
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case AQ_NORMAL: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY_DBLE_FRAMESIZE; break;
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case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE; break;
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}
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}
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}
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else /* CT_OPUS64 */
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{
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iOPUSFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
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if ( eAudioChannelConf == CC_MONO )
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{
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CurOpusEncoder = Opus64EncoderMono;
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CurOpusDecoder = Opus64DecoderMono;
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iNumAudioChannels = 1;
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switch ( eAudioQuality )
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switch ( eAudioQuality )
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{
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{
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case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_LOW_QUALITY; break;
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case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_LOW_QUALITY; break;
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@ -761,8 +790,8 @@ void CClient::Init()
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}
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}
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else
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else
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{
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{
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CurOpusEncoder = OpusEncoderStereo;
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CurOpusEncoder = Opus64EncoderStereo;
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CurOpusDecoder = OpusDecoderStereo;
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CurOpusDecoder = Opus64DecoderStereo;
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iNumAudioChannels = 2;
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iNumAudioChannels = 2;
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switch ( eAudioQuality )
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switch ( eAudioQuality )
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@ -772,6 +801,10 @@ void CClient::Init()
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case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY; break;
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case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY; break;
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}
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}
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}
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}
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}
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// calculate stereo (two channels) buffer size
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iStereoBlockSizeSam = 2 * iMonoBlockSizeSam;
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vecCeltData.Init ( iCeltNumCodedBytes );
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vecCeltData.Init ( iCeltNumCodedBytes );
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@ -789,6 +822,29 @@ void CClient::Init()
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iSndCrdFrameSizeFactor,
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iSndCrdFrameSizeFactor,
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iNumAudioChannels );
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iNumAudioChannels );
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// init reverberation
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AudioReverbL.Init ( SYSTEM_SAMPLE_RATE_HZ );
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AudioReverbR.Init ( SYSTEM_SAMPLE_RATE_HZ );
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// init the sound card conversion buffers
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if ( bSndCrdConversionBufferRequired )
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{
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// inits for conversion buffer (the size of the conversion buffer must
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// be the sum of input/output sizes which is the worst case fill level)
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const int iSndCardStereoBlockSizeSamConvBuff = 2 * iSndCardMonoBlockSizeSamConvBuff;
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const int iConBufSize = iStereoBlockSizeSam + iSndCardStereoBlockSizeSamConvBuff;
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SndCrdConversionBufferIn.Init ( iConBufSize );
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SndCrdConversionBufferOut.Init ( iConBufSize );
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vecDataConvBuf.Init ( iStereoBlockSizeSam );
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// the output conversion buffer must be filled with the inner
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// block size for initialization (this is the latency which is
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// introduced by the conversion buffer) to avoid buffer underruns
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const CVector<int16_t> vZeros ( iStereoBlockSizeSam, 0 );
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SndCrdConversionBufferOut.Put ( vZeros, vZeros.Size() );
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}
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// reset initialization phase flag and mute flag
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// reset initialization phase flag and mute flag
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bIsInitializationPhase = true;
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bIsInitializationPhase = true;
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bMuteInputAndOutput = false;
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bMuteInputAndOutput = false;
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16
src/client.h
16
src/client.h
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@ -83,23 +83,19 @@
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// Fs: sampling rate (SYSTEM_SAMPLE_RATE_HZ)
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// Fs: sampling rate (SYSTEM_SAMPLE_RATE_HZ)
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// L: number of samples per packet (SYSTEM_FRAME_SIZE_SAMPLES)
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// L: number of samples per packet (SYSTEM_FRAME_SIZE_SAMPLES)
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// N: number of bytes per packet (values below)
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// N: number of bytes per packet (values below)
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#if ( SYSTEM_FRAME_SIZE_SAMPLES == 64 )
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#define OPUS_NUM_BYTES_MONO_LOW_QUALITY 12
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#define OPUS_NUM_BYTES_MONO_LOW_QUALITY 12
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#define OPUS_NUM_BYTES_MONO_NORMAL_QUALITY 22
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#define OPUS_NUM_BYTES_MONO_NORMAL_QUALITY 22
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#define OPUS_NUM_BYTES_MONO_HIGH_QUALITY 35
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#define OPUS_NUM_BYTES_MONO_HIGH_QUALITY 35
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#define OPUS_NUM_BYTES_MONO_LOW_QUALITY_DBLE_FRAMESIZE 25
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#define OPUS_NUM_BYTES_MONO_NORMAL_QUALITY_DBLE_FRAMESIZE 45
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#define OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE 71
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#define OPUS_NUM_BYTES_STEREO_LOW_QUALITY 24
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#define OPUS_NUM_BYTES_STEREO_LOW_QUALITY 24
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#define OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY 35
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#define OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY 35
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#define OPUS_NUM_BYTES_STEREO_HIGH_QUALITY 72
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#define OPUS_NUM_BYTES_STEREO_HIGH_QUALITY 72
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#else
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#define OPUS_NUM_BYTES_STEREO_LOW_QUALITY_DBLE_FRAMESIZE 47
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# define OPUS_NUM_BYTES_MONO_LOW_QUALITY 25
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#define OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY_DBLE_FRAMESIZE 71
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# define OPUS_NUM_BYTES_MONO_NORMAL_QUALITY 45
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#define OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE 142
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# define OPUS_NUM_BYTES_MONO_HIGH_QUALITY 71
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# define OPUS_NUM_BYTES_STEREO_LOW_QUALITY 47
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# define OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY 71
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# define OPUS_NUM_BYTES_STEREO_HIGH_QUALITY 142
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#endif
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/* Classes ********************************************************************/
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/* Classes ********************************************************************/
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@ -128,21 +128,12 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
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"connection properties.<br>"
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"connection properties.<br>"
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"Three buffer sizes are supported:"
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"Three buffer sizes are supported:"
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"<ul>"
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"<ul>"
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#if ( SYSTEM_FRAME_SIZE_SAMPLES == 64 )
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"<li>64 samples: This is the preferred setting since it gives lowest "
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"<li>64 samples: This is the preferred setting since it gives lowest "
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"latency but does not work with all sound cards.</li>"
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"latency but does not work with all sound cards.</li>"
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"<li>128 samples: This setting should work on most of the available "
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"<li>128 samples: This setting should work on most of the available "
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"sound cards.</li>"
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"sound cards.</li>"
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"<li>256 samples: This setting should only be used if only a very slow "
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"<li>256 samples: This setting should only be used if only a very slow "
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"computer or a slow internet connection is available.</li>"
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"computer or a slow internet connection is available.</li>"
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#else
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"<li>128 samples: This is the preferred setting since it gives lowest "
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"latency but does not work with all sound cards.</li>"
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"<li>256 samples: This setting should work on most of the available "
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"sound cards.</li>"
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"<li>512 samples: This setting should only be used if only a very slow "
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"computer or a slow internet connection is available.</li>"
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#endif
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"</ul>"
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"</ul>"
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"Some sound card driver do not allow the buffer delay to be changed "
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"Some sound card driver do not allow the buffer delay to be changed "
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"from within the " ) + APP_NAME +
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"from within the " ) + APP_NAME +
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@ -356,14 +347,14 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
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// set text for sound card buffer delay radio buttons
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// set text for sound card buffer delay radio buttons
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rbtBufferDelayPreferred->setText ( GenSndCrdBufferDelayString (
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rbtBufferDelayPreferred->setText ( GenSndCrdBufferDelayString (
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FRAME_SIZE_FACTOR_PREFERRED * SYSTEM_FRAME_SIZE_SAMPLES,
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FRAME_SIZE_FACTOR_PREFERRED * SYSTEM_FRAME_SIZE_SAMPLES_SMALL,
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", preferred" ) );
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", preferred" ) );
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rbtBufferDelayDefault->setText ( GenSndCrdBufferDelayString (
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rbtBufferDelayDefault->setText ( GenSndCrdBufferDelayString (
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FRAME_SIZE_FACTOR_DEFAULT * SYSTEM_FRAME_SIZE_SAMPLES ) );
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FRAME_SIZE_FACTOR_DEFAULT * SYSTEM_FRAME_SIZE_SAMPLES_SMALL ) );
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rbtBufferDelaySafe->setText ( GenSndCrdBufferDelayString (
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rbtBufferDelaySafe->setText ( GenSndCrdBufferDelayString (
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FRAME_SIZE_FACTOR_SAFE * SYSTEM_FRAME_SIZE_SAMPLES ) );
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FRAME_SIZE_FACTOR_SAFE * SYSTEM_FRAME_SIZE_SAMPLES_SMALL ) );
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||||||
|
|
||||||
// sound card buffer delay inits
|
// sound card buffer delay inits
|
||||||
SndCrdBufferDelayButtonGroup.addButton ( rbtBufferDelayPreferred );
|
SndCrdBufferDelayButtonGroup.addButton ( rbtBufferDelayPreferred );
|
||||||
|
@ -478,18 +469,12 @@ QString CClientSettingsDlg::GenSndCrdBufferDelayString ( const int iFrameSiz
|
||||||
void CClientSettingsDlg::UpdateSoundCardFrame()
|
void CClientSettingsDlg::UpdateSoundCardFrame()
|
||||||
{
|
{
|
||||||
// get current actual buffer size value
|
// get current actual buffer size value
|
||||||
const int iCurActualBufSize =
|
const int iCurActualBufSize = pClient->GetSndCrdActualMonoBlSize();
|
||||||
pClient->GetSndCrdActualMonoBlSize();
|
|
||||||
|
|
||||||
// check which predefined size is used (it is possible that none is used)
|
// check which predefined size is used (it is possible that none is used)
|
||||||
const bool bPreferredChecked =
|
const bool bPreferredChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_PREFERRED );
|
||||||
( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED );
|
const bool bDefaultChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_DEFAULT );
|
||||||
|
const bool bSafeChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_SAFE );
|
||||||
const bool bDefaultChecked =
|
|
||||||
( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT );
|
|
||||||
|
|
||||||
const bool bSafeChecked =
|
|
||||||
( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE );
|
|
||||||
|
|
||||||
// Set radio buttons according to current value (To make it possible
|
// Set radio buttons according to current value (To make it possible
|
||||||
// to have all radio buttons unchecked, we have to disable the
|
// to have all radio buttons unchecked, we have to disable the
|
||||||
|
@ -504,14 +489,9 @@ void CClientSettingsDlg::UpdateSoundCardFrame()
|
||||||
SndCrdBufferDelayButtonGroup.setExclusive ( true );
|
SndCrdBufferDelayButtonGroup.setExclusive ( true );
|
||||||
|
|
||||||
// disable radio buttons which are not supported by audio interface
|
// disable radio buttons which are not supported by audio interface
|
||||||
rbtBufferDelayPreferred->setEnabled (
|
rbtBufferDelayPreferred->setEnabled ( pClient->GetFraSiFactPrefSupported() );
|
||||||
pClient->GetFraSiFactPrefSupported() );
|
rbtBufferDelayDefault->setEnabled ( pClient->GetFraSiFactDefSupported() );
|
||||||
|
rbtBufferDelaySafe->setEnabled ( pClient->GetFraSiFactSafeSupported() );
|
||||||
rbtBufferDelayDefault->setEnabled (
|
|
||||||
pClient->GetFraSiFactDefSupported() );
|
|
||||||
|
|
||||||
rbtBufferDelaySafe->setEnabled (
|
|
||||||
pClient->GetFraSiFactSafeSupported() );
|
|
||||||
|
|
||||||
// If any of our predefined sizes is chosen, use the regular group box
|
// If any of our predefined sizes is chosen, use the regular group box
|
||||||
// title text. If not, show the actual buffer size. Otherwise the user
|
// title text. If not, show the actual buffer size. Otherwise the user
|
||||||
|
|
Loading…
Reference in a new issue