small changes
This commit is contained in:
Volker Fischer
parent
068b78fb24
commit
ceacfd9a13
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@ -132,7 +132,7 @@ void CClient::OnProtocolStatus ( bool bOk )
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void CClient::Init()
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{
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// set block sizes (in samples)
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iBlockSizeSam = MIN_BLOCK_SIZE_SAMPLES;
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iBlockSizeSam = MIN_BLOCK_SIZE_SAMPLES;
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iSndCrdBlockSizeSam = MIN_SND_CRD_BLOCK_SIZE_SAMPLES;
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vecsAudioSndCrd.Init ( iSndCrdBlockSizeSam * 2 ); // stereo
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@ -70,8 +70,8 @@ public:
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double MicLevelR() { return SignalLevelMeterR.MicLevel(); }
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bool IsConnected() { return Channel.IsConnected(); }
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/* we want to return the standard deviation. For that we need to calculate
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the sqaure root */
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/* We want to return the standard deviation. For that we need to calculate
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the sqaure root. */
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double GetTimingStdDev() { return sqrt ( RespTimeMoAvBuf.GetAverage() ); }
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int GetAudioInFader() { return iAudioInFader; }
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22
src/global.h
22
src/global.h
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@ -1,5 +1,5 @@
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/******************************************************************************\
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* Copyright (c) 2004-2006
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* Copyright (c) 2004-2008
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*
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* Author(s):
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* Volker Fischer
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@ -11,13 +11,13 @@
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* Foundation; either version 2 of the License, or (at your option) any later
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* version.
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*
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
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* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
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* details.
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*
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* You should have received a copy of the GNU General Public License along with
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* this program; if not, write to the Free Software Foundation, Inc.,
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* this program; if not, write to the Free Software Foundation, Inc.,
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* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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\******************************************************************************/
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@ -49,19 +49,19 @@
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// file name for logging file
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#define LOG_FILE_NAME "llconsrvlog.txt"
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/* defined port number for client and server */
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// defined port number for client and server
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#define LLCON_PORT_NUMBER 22122
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/* sample rate */
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// sample rate
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#define SAMPLE_RATE 24000
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/* sound card sample rate. Should be always 48 kHz to avoid sound card driver
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internal sample rate conversion which might be buggy */
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// sound card sample rate. Should be always 48 kHz to avoid sound card driver
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// internal sample rate conversion which might be buggy
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#define SND_CRD_SAMPLE_RATE 48000
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/* minimum block duration - all other buffer durations must be a multiple
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of this duration */
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#define MIN_BLOCK_DURATION_MS 2 /* ms */
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// minimum block duration - all other buffer durations must be a multiple
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// of this duration
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#define MIN_BLOCK_DURATION_MS 2 // ms
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#define MIN_BLOCK_SIZE_SAMPLES ( MIN_BLOCK_DURATION_MS * SAMPLE_RATE / 1000 )
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#define MIN_SND_CRD_BLOCK_SIZE_SAMPLES ( MIN_BLOCK_DURATION_MS * SND_CRD_SAMPLE_RATE / 1000 )
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@ -1,5 +1,5 @@
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/******************************************************************************\
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* Copyright (c) 2004-2007
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* Copyright (c) 2004-2008
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*
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* Author(s):
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* Volker Fischer
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@ -11,13 +11,13 @@
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* Foundation; either version 2 of the License, or (at your option) any later
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* version.
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*
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
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* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
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* details.
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*
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* You should have received a copy of the GNU General Public License along with
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* this program; if not, write to the Free Software Foundation, Inc.,
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* this program; if not, write to the Free Software Foundation, Inc.,
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* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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\******************************************************************************/
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@ -1,5 +1,5 @@
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/******************************************************************************\
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* Copyright (c) 2004-2006
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* Copyright (c) 2004-2008
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*
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* Author(s):
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* Volker Fischer
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@ -332,6 +332,22 @@ void CSound::InitRecordingAndPlayback ( int iNewBufferSize )
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// TODO this should be done in the setinoutbuf functions
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// create and activate buffers
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ASIOCreateBuffers(bufferInfos, 2 * NUM_IN_OUT_CHANNELS,
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iBufferSize * BYTES_PER_SAMPLE, &asioCallbacks);
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// now set all the buffer details
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for ( i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
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{
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channelInfos[i].channel = NUM_IN_OUT_CHANNELS;
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channelInfos[i].isInput = bufferInfos[i].isInput;
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ASIOGetChannelInfo ( &channelInfos[i] );
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}
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/*
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// reset interface so that all buffers are returned from the interface
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waveInReset ( m_WaveIn );
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@ -380,20 +396,7 @@ void CSound::InitRecordingAndPlayback ( int iNewBufferSize )
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// TODO this should be done in the setinoutbuf functions
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// create and activate buffers
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// ASIOCreateBuffers(bufferInfos, 2 * NUM_IN_OUT_CHANNELS,
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// iBufferSizeIn * BYTES_PER_SAMPLE, &asioCallbacks);
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ASIOCreateBuffers(bufferInfos, 2 * NUM_IN_OUT_CHANNELS,
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iBufferSize * BYTES_PER_SAMPLE, &asioCallbacks);
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// now set all the buffer details
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for ( i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
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{
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channelInfos[i].channel = NUM_IN_OUT_CHANNELS;
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channelInfos[i].isInput = bufferInfos[i].isInput;
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ASIOGetChannelInfo ( &channelInfos[i] );
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}
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@ -519,7 +522,7 @@ pstrDevices[0] = driverInfo.name;
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long lNumInChan;
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long lNumOutChan;
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ASIOGetChannels ( &lNumInChan, &lNumOutChan );
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if ( ( lNumInChan != NUM_IN_OUT_CHANNELS ) || ( lNumOutChan != NUM_IN_OUT_CHANNELS ) )
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if ( ( lNumInChan < NUM_IN_OUT_CHANNELS ) || ( lNumOutChan < NUM_IN_OUT_CHANNELS ) )
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{
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throw CGenErr ( "The audio device does not support required number of channels." );
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}
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@ -641,7 +644,7 @@ ASIOTime* CSound::bufferSwitchTimeInfo ( ASIOTime *timeInfo, long index, ASIOBoo
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return 0L;
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}
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void CSound::bufferSwitch( long index, ASIOBool processNow )
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void CSound::bufferSwitch ( long index, ASIOBool processNow )
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{
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@ -861,7 +864,7 @@ void CSound::InitRecording ( int iNewBufferSize, bool bNewBlocking )
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// set internal parameter
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iBufferSizeIn = iNewBufferSize;
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bBlockingRec = bNewBlocking;
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bBlockingRec = bNewBlocking;
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// reset interface so that all buffers are returned from the interface
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waveInReset ( m_WaveIn );
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@ -35,8 +35,8 @@
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/* Definitions ****************************************************************/
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// switch here between ASIO (Steinberg) or native Windows(TM) sound interface
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#undef USE_ASIO_SND_INTERFACE
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//#define USE_ASIO_SND_INTERFACE
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//#undef USE_ASIO_SND_INTERFACE
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#define USE_ASIO_SND_INTERFACE
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#define NUM_IN_OUT_CHANNELS 2 /* Stereo recording (but we only
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