bug fix in socket buffer, fix in channel, ASIO buffer size setting should work now, some other small fixes

src/buffer.cpp

@@ -380,10 +380,13 @@ void CNetBuf::Clear ( const EClearType eClearType )
 
 void CNetBuf::FadeInAudioDataBlock ( CVector<double>& vecdData )
 {
+    // correct fading length if necessary
+    const int iCurFadingLen = min ( vecdData.Size(), iNumSamFading );
+
     // apply linear fading
-    for ( int i = 0; i < iNumSamFading; i++ )
+    for ( int i = 0; i < iCurFadingLen; i++ )
     {
-        vecdData[i] *= ( (double) i / iNumSamFading );
+        vecdData[i] *= ( (double) i / iCurFadingLen );
     }
 
     // reset flag
@@ -394,14 +397,17 @@ void CNetBuf::FadeOutExtrapolateAudioDataBlock ( CVector<double>& vecdData,
                                                  const double dExPDiff,
                                                  const double dExPLastV )
 {
+    // correct fading length if necessary
+    const int iCurFadingLenExtra = min ( vecdData.Size(), iNumSamFadingExtra );
+
     // apply linear extrapolation and linear fading
-    for ( int i = 0; i < iNumSamFadingExtra; i++ )
+    for ( int i = 0; i < iCurFadingLenExtra; i++ )
     {
         // calculate extrapolated value
         vecdData[i] = ( ( i + 1 ) * dExPDiff + dExPLastV );
 
         // linear fading
-        vecdData[i] *= ( (double) ( iNumSamFadingExtra - i ) / iNumSamFadingExtra );
+        vecdData[i] *= ( (double) ( iCurFadingLenExtra - i ) / iCurFadingLenExtra );
     }
 }
 

src/channel.cpp

@@ -622,11 +622,19 @@ CChannel::CChannel ( const bool bNIsServer ) : bIsServer ( bNIsServer ),
         MIN_SERVER_BLOCK_SIZE_SAMPLES * DEF_NET_BLOCK_SIZE_FACTOR,
         CT_MSADPCM );
 
+    if ( bIsServer )
+    {
         SetNetwBufSizeFactOut ( DEF_NET_BLOCK_SIZE_FACTOR );
+    }
+    else
+    {
+        SetNetwBufSizeOut ( MIN_SERVER_BLOCK_SIZE_SAMPLES );
+    }
 
     // set initial audio compression format for output
     SetAudioCompressionOut ( CT_MSADPCM );
 
+
     // connections -------------------------------------------------------------
     QObject::connect ( &Protocol,
         SIGNAL ( MessReadyForSending ( CVector<uint8_t> ) ),
@@ -730,12 +738,17 @@ void CChannel::SetNetwBufSizeOut ( const int iNewAudioBlockSizeOut )
     // this function is intended for the client (not the server)
     QMutexLocker locker ( &Mutex );
 
+    // direct setting of audio buffer (without buffer size factor) is
+    // right now only intendet for the client, not the server
+    if ( !bIsServer )
+    {
         // store new value
         iCurAudioBlockSizeOut = iNewAudioBlockSizeOut;
 
         iAudComprSizeOut =
             AudioCompressionOut.Init ( iNewAudioBlockSizeOut, eAudComprTypeOut );
     }
+}
 
 void CChannel::SetNetwBufSizeFactOut ( const int iNewNetwBlSiFactOut )
 {

src/client.cpp

@@ -150,9 +150,6 @@ bool CClient::SetServerAddr ( QString strNAddr )
     return true;
 }
 
-
-
-
 void CClient::SetSndCrdPreferredMonoBlSizeIndex ( const int iNewIdx )
 {
     // right now we simply set the internal value
@@ -161,18 +158,32 @@ if ( ( iNewIdx >= 0 ) && ( CSndCrdBufferSizes::GetNumOfBufferSizes() ) )
         iSndCrdPreferredMonoBlSizeIndex = iNewIdx;
     }
 
-// TODO take action on new parameter
+    // init with new parameter, if client was running then first
+    // stop it and restart again after new initialization
+    const bool bWasRunning = Sound.IsRunning();
+    if ( bWasRunning )
+    {
+        Sound.Stop();
     }
 
+    // init with new block size index parameter
+    Init ( iSndCrdPreferredMonoBlSizeIndex );
 
+    if ( bWasRunning )
+    {
+        Sound.Start();
+    }
 
+    // tell the server that audio coding has changed (it
+    // is important to call this function AFTER we have applied
+    // the new setting to the channel!)
+    Channel.CreateNetTranspPropsMessFromCurrentSettings();
+}
 
 void CClient::Start()
 {
     // init object
-
+    Init ( iSndCrdPreferredMonoBlSizeIndex );
-// TEST
-Init ( 192 );
 
     // enable channel
     Channel.SetEnable ( true );
@@ -203,8 +214,12 @@ void CClient::AudioCallback ( CVector<short>& psData, void* arg )
     pMyClientObj->ProcessAudioData ( psData );
 }
 
-void CClient::Init ( const int iPrefMonoBlockSizeSamAtSndCrdSamRate )
+void CClient::Init ( const int iPrefMonoBlockSizeSamIndexAtSndCrdSamRate )
 {
+    // translate block size index in actual block size
+    const int iPrefMonoBlockSizeSamAtSndCrdSamRate = CSndCrdBufferSizes::
+        GetBufferSizeFromIndex ( iPrefMonoBlockSizeSamIndexAtSndCrdSamRate );
+
     // get actual sound card buffer size using preferred size
     iSndCrdMonoBlockSizeSam   = Sound.Init ( iPrefMonoBlockSizeSamAtSndCrdSamRate );
     iSndCrdStereoBlockSizeSam = 2 * iSndCrdMonoBlockSizeSam;

src/client.h

@@ -156,7 +156,7 @@ protected:
     // callback function must be static, otherwise it does not work
     static void  AudioCallback ( CVector<short>& psData, void* arg );
 
-    void         Init ( const int iPrefMonoBlockSizeSamAtSndCrdSamRate );
+    void         Init ( const int iPrefMonoBlockSizeSamIndexAtSndCrdSamRate );
     void         ProcessAudioData ( CVector<short>& vecsStereoSndCrd );
     void         UpdateTimeResponseMeasurement();
     void         UpdateSocketBufferSize();

src/clientsettingsdlgbase.ui

@@ -251,7 +251,7 @@
              <property name="minimumSize" >
               <size>
                <width>110</width>
-               <height>0</height>
+               <height>20</height>
               </size>
              </property>
              <property name="frameShape" >
@@ -281,7 +281,7 @@
              <property name="minimumSize" >
               <size>
                <width>110</width>
-               <height>0</height>
+               <height>20</height>
               </size>
              </property>
              <property name="frameShape" >
@@ -386,7 +386,7 @@
           <property name="minimumSize" >
            <size>
             <width>50</width>
-            <height>0</height>
+            <height>20</height>
            </size>
           </property>
           <property name="frameShape" >
@@ -427,7 +427,7 @@
             <property name="minimumSize" >
              <size>
               <width>50</width>
-              <height>0</height>
+              <height>20</height>
              </size>
             </property>
             <property name="font" >
@@ -484,6 +484,12 @@
         </item>
         <item>
          <widget class="QLabel" name="TextUpstreamValue" >
+          <property name="minimumSize" >
+           <size>
+            <width>0</width>
+            <height>20</height>
+           </size>
+          </property>
           <property name="font" >
            <font>
             <weight>75</weight>

src/soundbase.cpp

@@ -61,11 +61,6 @@ void CSoundBase::Stop()
     {
         wait ( 5000 );
     }
-
-
-// TODO stop audio interface (previously done in Close function, we
-// better should implement a stop function in derived sound classes
-    Close();
 }
 
 void CSoundBase::run()

windows/sound.cpp

@@ -43,9 +43,8 @@ ASIOBufferInfo   bufferInfos[2 * NUM_IN_OUT_CHANNELS]; // for input and output b
 ASIOChannelInfo  channelInfos[2 * NUM_IN_OUT_CHANNELS];
 bool             bASIOPostOutput;
 ASIOCallbacks    asioCallbacks;
-int              iBufferSizeMono;
-int              iBufferSizeStereo;
 int              iASIOBufferSizeMono;
+int              iASIOBufferSizeStereo;
 
 CVector<short>   vecsTmpAudioSndCrdStereo;
 
@@ -88,12 +87,12 @@ void CSound::SetDev ( const int iNewDev )
             // loading and initializing the new driver failed, go back to original
             // driver and display error message
             LoadAndInitializeDriver ( lCurDev );
-            Init ( iBufferSizeStereo );
+            Init ( iASIOBufferSizeStereo );
 
             throw CGenErr ( strErrorMessage.c_str() );
         }
 
-        Init ( iBufferSizeStereo );
+        Init ( iASIOBufferSizeStereo );
     }
     else
     {
@@ -319,92 +318,75 @@ int CSound::GetActualBufferSize ( const int iDesiredBufferSizeMono )
 
 int CSound::Init ( const int iNewPrefMonoBufferSize )
 {
-    // first, stop audio and dispose ASIO buffers
-    ASIOStop();
-
     ASIOMutex.lock(); // get mutex lock
     {
+        // get the actual sound card buffer size which is supported
+        // by the audio hardware
+        iASIOBufferSizeMono =
+            GetActualBufferSize ( iNewPrefMonoBufferSize );
+
         // init base clasee
-        CSoundBase::Init ( iNewPrefMonoBufferSize );
+        CSoundBase::Init ( iASIOBufferSizeMono );
 
         // set internal buffer size value and calculate stereo buffer size
-        iBufferSizeMono   = iNewPrefMonoBufferSize;
+        iASIOBufferSizeStereo = 2 * iASIOBufferSizeMono;
-        iBufferSizeStereo = 2 * iBufferSizeMono;
 
-
+        // set the sample rate
-// TODO possible BUG!!!!!!!!!!!!!!!!!!!!!
-// iBufferSizeMono must not be the same as iASIOBufferSizeMono
-
-
-        // create memory for intermediate audio buffer
-        vecsTmpAudioSndCrdStereo.Init ( iBufferSizeStereo );
-
-
-
-
-    int i;
-
-    // set the sample rate and check if sample rate is supported
         ASIOSetSampleRate ( SND_CRD_SAMPLE_RATE );
 
-
+        // create memory for intermediate audio buffer
-// TEST
+        vecsTmpAudioSndCrdStereo.Init ( iASIOBufferSizeStereo );
-iASIOBufferSizeMono = GetActualBufferSize ( iBufferSizeMono );
-
-
-    for ( i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
-    {
-        // prepare input channels
-        bufferInfos[i].isInput    = ASIOTrue;
-        bufferInfos[i].channelNum = i;
-        bufferInfos[i].buffers[0] = 0;
-        bufferInfos[i].buffers[1] = 0;
-
-        // prepare output channels
-        bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput    = ASIOFalse;
-        bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = i;
-        bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
-        bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
-    }
 
         // create and activate ASIO buffers (buffer size in samples)
-    ASIOCreateBuffers ( bufferInfos, 2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
+        ASIOCreateBuffers ( bufferInfos,
+            2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
             iASIOBufferSizeMono, &asioCallbacks );
 
-
-
         // check wether the driver requires the ASIOOutputReady() optimization
         // (can be used by the driver to reduce output latency by one block)
         bASIOPostOutput = ( ASIOOutputReady() == ASE_OK );
-
-
-
     }
     ASIOMutex.unlock();
 
-    // initialization is done, (re)start audio
+    return iASIOBufferSizeMono;
+}
+
+void CSound::Start()
+{
+    // start audio
     ASIOStart();
 
-// TEST
+    // call base class
-return iNewPrefMonoBufferSize;
+    CSoundBase::Start();
+}
+
+void CSound::Stop()
+{
+    // stop audio
+    ASIOStop();
+
+    // call base class
+    CSoundBase::Stop();
 }
 
 void CSound::Close()
 {
-    // stop driver
+
-    ASIOStop();
+// TODO
+
 }
 
 CSound::CSound ( void (*fpNewCallback) ( CVector<short>& psData, void* arg ), void* arg ) :
     CSoundBase ( true, fpNewCallback, arg )
 {
+    int i;
 
 // TEST
 pSound = this;
 
 
     // get available ASIO driver names in system
-    for ( int i = 0; i < MAX_NUMBER_SOUND_CARDS; i++ )
+    for ( i = 0; i < MAX_NUMBER_SOUND_CARDS; i++ )
     {
         cDriverNames[i] = new char[32];
     }
@@ -422,6 +404,23 @@ pSound = this;
     // init device index with illegal value to show that driver is not initialized
     lCurDev = -1;
 
+    // init buffer infos, we always want to have two input and
+    // two output channels
+    for ( i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
+    {
+        // prepare input channels
+        bufferInfos[i].isInput    = ASIOTrue;
+        bufferInfos[i].channelNum = i;
+        bufferInfos[i].buffers[0] = 0;
+        bufferInfos[i].buffers[1] = 0;
+
+        // prepare output channels
+        bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput    = ASIOFalse;
+        bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = i;
+        bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
+        bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
+    }
+
     // set up the asioCallback structure
     asioCallbacks.bufferSwitch         = &bufferSwitch;
     asioCallbacks.sampleRateDidChange  = &sampleRateChanged;

windows/sound.h

@@ -60,6 +60,8 @@ public:
     virtual ~CSound();
 
     virtual int  Init ( const int iNewPrefMonoBufferSize );
+    virtual void Start();
+    virtual void Stop();
     virtual void Close();
 
     virtual void OpenDriverSetup() { ASIOControlPanel(); }