small changes

src/client.cpp

@@ -132,7 +132,7 @@ void CClient::OnProtocolStatus ( bool bOk )
 void CClient::Init()
 {
     // set block sizes (in samples)
-    iBlockSizeSam = MIN_BLOCK_SIZE_SAMPLES;
+    iBlockSizeSam       = MIN_BLOCK_SIZE_SAMPLES;
     iSndCrdBlockSizeSam = MIN_SND_CRD_BLOCK_SIZE_SAMPLES;
 
     vecsAudioSndCrd.Init  ( iSndCrdBlockSizeSam * 2 ); // stereo

src/client.h

@@ -70,8 +70,8 @@ public:
     double MicLevelR() { return SignalLevelMeterR.MicLevel(); }
     bool   IsConnected() { return Channel.IsConnected(); }
 
-    /* we want to return the standard deviation. For that we need to calculate
+    /* We want to return the standard deviation. For that we need to calculate
-       the sqaure root */
+       the sqaure root. */
     double GetTimingStdDev() { return sqrt ( RespTimeMoAvBuf.GetAverage() ); }
 
     int GetAudioInFader() { return iAudioInFader; }

src/global.h

@@ -1,5 +1,5 @@
 /******************************************************************************\
- * Copyright (c) 2004-2006
+ * Copyright (c) 2004-2008
  *
  * Author(s):
  *  Volker Fischer
@@ -49,19 +49,19 @@
 // file name for logging file
 #define LOG_FILE_NAME                   "llconsrvlog.txt"
 
-/* defined port number for client and server */
+// defined port number for client and server
 #define LLCON_PORT_NUMBER               22122
 
-/* sample rate */
+// sample rate
 #define SAMPLE_RATE                     24000
 
-/* sound card sample rate. Should be always 48 kHz to avoid sound card driver
+// sound card sample rate. Should be always 48 kHz to avoid sound card driver
-   internal sample rate conversion which might be buggy */
+// internal sample rate conversion which might be buggy
 #define SND_CRD_SAMPLE_RATE             48000
 
-/* minimum block duration - all other buffer durations must be a multiple
+// minimum block duration - all other buffer durations must be a multiple
-   of this duration */
+// of this duration
-#define MIN_BLOCK_DURATION_MS           2 /* ms */
+#define MIN_BLOCK_DURATION_MS           2 // ms
 
 #define MIN_BLOCK_SIZE_SAMPLES          ( MIN_BLOCK_DURATION_MS * SAMPLE_RATE / 1000 )
 #define MIN_SND_CRD_BLOCK_SIZE_SAMPLES  ( MIN_BLOCK_DURATION_MS * SND_CRD_SAMPLE_RATE / 1000 )

src/settings.cpp

@@ -1,5 +1,5 @@
 /******************************************************************************\
- * Copyright (c) 2004-2007
+ * Copyright (c) 2004-2008
  *
  * Author(s):
  *  Volker Fischer

windows/sound.cpp

@@ -1,5 +1,5 @@
 /******************************************************************************\
- * Copyright (c) 2004-2006
+ * Copyright (c) 2004-2008
  *
  * Author(s):
  *  Volker Fischer
@@ -332,6 +332,22 @@ void CSound::InitRecordingAndPlayback ( int iNewBufferSize )
 
 
 
+// TODO this should be done in the setinoutbuf functions
+	// create and activate buffers
+	ASIOCreateBuffers(bufferInfos, 2 * NUM_IN_OUT_CHANNELS,
+		iBufferSize * BYTES_PER_SAMPLE, &asioCallbacks);
+
+	// now set all the buffer details
+	for ( i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
+	{
+		channelInfos[i].channel = NUM_IN_OUT_CHANNELS;
+		channelInfos[i].isInput = bufferInfos[i].isInput;
+		ASIOGetChannelInfo ( &channelInfos[i] );
+	}
+
+
+
+
 /*
     // reset interface so that all buffers are returned from the interface
     waveInReset ( m_WaveIn );
@@ -380,20 +396,7 @@ void CSound::InitRecordingAndPlayback ( int iNewBufferSize )
 
 
 
-// TODO this should be done in the setinoutbuf functions
-	// create and activate buffers
-//	ASIOCreateBuffers(bufferInfos, 2 * NUM_IN_OUT_CHANNELS,
-//		iBufferSizeIn * BYTES_PER_SAMPLE, &asioCallbacks);
-	ASIOCreateBuffers(bufferInfos, 2 * NUM_IN_OUT_CHANNELS,
-		iBufferSize * BYTES_PER_SAMPLE, &asioCallbacks);
 
-	// now set all the buffer details
-	for ( i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
-	{
-		channelInfos[i].channel = NUM_IN_OUT_CHANNELS;
-		channelInfos[i].isInput = bufferInfos[i].isInput;
-		ASIOGetChannelInfo ( &channelInfos[i] );
-	}
 
 
 
@@ -519,7 +522,7 @@ pstrDevices[0] = driverInfo.name;
 	long lNumInChan;
 	long lNumOutChan;
 	ASIOGetChannels ( &lNumInChan, &lNumOutChan );
-    if ( ( lNumInChan != NUM_IN_OUT_CHANNELS ) || ( lNumOutChan != NUM_IN_OUT_CHANNELS ) )
+    if ( ( lNumInChan < NUM_IN_OUT_CHANNELS ) || ( lNumOutChan < NUM_IN_OUT_CHANNELS ) )
     {
         throw CGenErr ( "The audio device does not support required number of channels." );
     }
@@ -641,7 +644,7 @@ ASIOTime* CSound::bufferSwitchTimeInfo ( ASIOTime *timeInfo, long index, ASIOBoo
 	return 0L;
 }
 
-void CSound::bufferSwitch( long index, ASIOBool processNow )
+void CSound::bufferSwitch ( long index, ASIOBool processNow )
 {
 
 
@@ -861,7 +864,7 @@ void CSound::InitRecording ( int iNewBufferSize, bool bNewBlocking )
 
     // set internal parameter
     iBufferSizeIn = iNewBufferSize;
-    bBlockingRec = bNewBlocking;
+    bBlockingRec  = bNewBlocking;
 
     // reset interface so that all buffers are returned from the interface
     waveInReset ( m_WaveIn );

windows/sound.h

@@ -35,8 +35,8 @@
 
 /* Definitions ****************************************************************/
 // switch here between ASIO (Steinberg) or native Windows(TM) sound interface
-#undef USE_ASIO_SND_INTERFACE
+//#undef USE_ASIO_SND_INTERFACE
-//#define USE_ASIO_SND_INTERFACE
+#define USE_ASIO_SND_INTERFACE
 
 
 #define NUM_IN_OUT_CHANNELS     2       /* Stereo recording (but we only