preparations for allowing different network block sizes

src/audiocompr.cpp

@@ -40,8 +40,12 @@ int CAudioCompression::Init(const int iNewAudioLen,
 
 	switch ( eNewAuCoTy )
 	{
-	case CT_NONE: return iCodeSize = 2 * iNewAudioLen; /* short = 2 * byte */
+	case CT_NONE:
-	case CT_IMAADPCM: return ImaAdpcm.Init(iNewAudioLen);
+		return iCodeSize = 2 * iNewAudioLen; /* short = 2 * byte */
+
+	case CT_IMAADPCM:
+		return ImaAdpcm.Init ( iNewAudioLen );
+
 	default: return 0;
 	}
 }
@@ -65,8 +69,11 @@ CVector<unsigned char> CAudioCompression::Encode(const CVector<short>& vecsAudio
 	{
 		switch ( eAudComprType )
 		{
-		case CT_IMAADPCM: return ImaAdpcm.Encode(vecsAudio); /* IMA-ADPCM */
+		case CT_IMAADPCM:
-		default: return CVector<unsigned char>(0);
+			return ImaAdpcm.Encode ( vecsAudio ); /* IMA-ADPCM */
+
+		default:
+			return CVector<unsigned char> ( 0 );
 		}
 	}
 }
@@ -93,8 +100,11 @@ CVector<short> CAudioCompression::Decode(const CVector<unsigned char>& vecbyAdpc
 	{
 		switch ( eAudComprType )
 		{
-		case CT_IMAADPCM: return ImaAdpcm.Decode(vecbyAdpcm); /* IMA-ADPCM */
+		case CT_IMAADPCM:
-		default: return CVector<short>(0);
+			return ImaAdpcm.Decode ( vecbyAdpcm ); /* IMA-ADPCM */
+
+		default:
+			return CVector<short> ( 0 );
 		}
 	}
 }

src/audiocompr.h

@@ -72,9 +72,14 @@ protected:
 	inline int CheckBounds ( const int iData, const int iMin, const int iMax )
 	{
 		if ( iData > iMax )
+		{
 			return iMax;
+		}
+
 		if ( iData < iMin )
+		{
 			return iMin;
+		}
 
 		return iData;
 	}
@@ -90,7 +95,8 @@ public:
 	CAudioCompression() {}
 	virtual ~CAudioCompression() {}
 
-	int Init(const int iNewAudioLen, const EAudComprType eNewAuCoTy);
+	int						Init ( const int iNewAudioLen,
+								   const EAudComprType eNewAuCoTy );
 	CVector<unsigned char>	Encode ( const CVector<short>& vecsAudio );
 	CVector<short>			Decode ( const CVector<unsigned char>& vecbyAdpcm );
 

src/channel.cpp

@@ -231,9 +231,9 @@ CChannel::CChannel ()
 	/* init conversion buffer */
 	ConvBuf.Init ( BLOCK_SIZE_SAMPLES );
 
-	/* init audio compression unit */
+	// set initial input and output block size factors
-	iAudComprSize = AudioCompression.Init ( BLOCK_SIZE_SAMPLES,
+	SetNetwInBlSiFact ( NET_BLOCK_SIZE_FACTOR );
-		CAudioCompression::CT_IMAADPCM );
+	SetNetwOutBlSiFact ( NET_BLOCK_SIZE_FACTOR );
 
 	/* init time-out for the buffer with zero -> no connection */
 	iConTimeOut = 0;
@@ -251,6 +251,32 @@ CChannel::CChannel ()
 		SIGNAL ( ReqJittBufSize() ) );
 }
 
+void CChannel::SetNetwInBlSiFact ( const int iNewBlockSizeFactor )
+{
+	// store new value
+	iCurNetwInBlSiFact = iNewBlockSizeFactor;
+
+	/* init audio compression unit */
+	iAudComprSizeIn = AudioCompressionIn.Init (
+		iNewBlockSizeFactor * MIN_BLOCK_SIZE_SAMPLES,
+		CAudioCompression::CT_IMAADPCM );
+
+	// initial value for connection time out counter
+	iConTimeOutStartVal = ( CON_TIME_OUT_SEC_MAX * 1000 ) /
+		( iNewBlockSizeFactor * MIN_BLOCK_SIZE_SAMPLES );
+}
+
+void CChannel::SetNetwOutBlSiFact ( const int iNewBlockSizeFactor )
+{
+	/* init audio compression unit */
+	iAudComprSizeOut = AudioCompressionOut.Init (
+		iNewBlockSizeFactor * MIN_BLOCK_SIZE_SAMPLES,
+		CAudioCompression::CT_IMAADPCM );
+
+	/* init conversion buffer */
+	ConvBuf.Init ( iNewBlockSizeFactor * MIN_BLOCK_SIZE_SAMPLES );
+}
+
 void CChannel::OnSendProtMessage ( CVector<uint8_t> vecMessage )
 {
 	// only send messages if we are connected, otherwise delete complete queue
@@ -304,11 +330,10 @@ EPutDataStat CChannel::PutData ( const CVector<unsigned char>& vecbyData,
 	EPutDataStat eRet = PS_GEN_ERROR;
 
 	/* only process if packet has correct size */
-	if ( iNumBytes == iAudComprSize )
+	if ( iNumBytes == iAudComprSizeIn )
 	{
 		/* decompress audio */
-		CVector<short> vecsDecomprAudio ( BLOCK_SIZE_SAMPLES );
+		CVector<short> vecsDecomprAudio ( AudioCompressionIn.Decode ( vecbyData ) );
-		vecsDecomprAudio = AudioCompression.Decode ( vecbyData );
 
 		/* do resampling to compensate for sample rate offsets in the
 		   different sound cards of the clients */
@@ -320,9 +345,10 @@ const int iInSize = ResampleObj.Resample(vecdResInData, vecdResOutData,
 	(double) SAMPLE_RATE / (SAMPLE_RATE - dSamRateOffset));
 */
 
-vecdResOutData.Init(BLOCK_SIZE_SAMPLES);
+vecdResOutData.Init ( iCurNetwInBlSiFact * MIN_BLOCK_SIZE_SAMPLES );
-for (int i = 0; i < BLOCK_SIZE_SAMPLES; i++)
+for ( int i = 0; i < iCurNetwInBlSiFact * MIN_BLOCK_SIZE_SAMPLES; i++ ) {
 	vecdResOutData[i] = (double) vecsDecomprAudio[i];
+}
 
 
 		Mutex.lock (); /* put mutex lock */
@@ -342,7 +368,7 @@ for (int i = 0; i < BLOCK_SIZE_SAMPLES; i++)
 		const bool bChanWasNotConnected = !IsConnected();
 
 		// reset time-out counter
-		iConTimeOut = CON_TIME_OUT_CNT_MAX;
+		iConTimeOut = iConTimeOutStartVal;
 
 		// if channel was not connected, emit signal to inform that new
 		// connection was established
@@ -404,8 +430,8 @@ CVector<unsigned char> CChannel::PrepSendPacket(const CVector<short>& vecsNPacke
 	if ( ConvBuf.Put ( vecsNPacket ) )
 	{
 		/* a packet is ready, compress audio */
-		vecbySendBuf.Init ( iAudComprSize );
+		vecbySendBuf.Init ( iAudComprSizeOut );
-		vecbySendBuf = AudioCompression.Encode ( ConvBuf.Get () );
+		vecbySendBuf = AudioCompressionOut.Encode ( ConvBuf.Get () );
 	}
 
 	return vecbySendBuf;

src/channel.h

@@ -40,7 +40,6 @@
    connected to not-connected (the actual time depends on the way the error
    correction is implemented) */
 #define CON_TIME_OUT_SEC_MAX	5 // seconds
-#define CON_TIME_OUT_CNT_MAX	( ( CON_TIME_OUT_SEC_MAX * 1000 ) / BLOCK_DURATION_MS )
 
 /* maximum number of internet connections (channels) */
 // if you want to change this paramter, change the connections in this class, too!
@@ -77,8 +76,6 @@ public:
 
 	bool IsConnected() const { return iConTimeOut > 0; }
 
-	int	GetComprAudSize() { return iAudComprSize; }
-
 	void SetAddress ( const CHostAddress NAddr ) { InetAddr = NAddr; }
 	bool GetAddress ( CHostAddress& RetAddr );
 	CHostAddress GetAddress () { return InetAddr; }
@@ -98,9 +95,14 @@ public:
 	void CreateReqJitBufMes() { Protocol.CreateReqJitBufMes(); }
 
 protected:
+	void SetNetwInBlSiFact ( const int iNewBlockSizeFactor );
+	void SetNetwOutBlSiFact ( const int iNewBlockSizeFactor );
+
 	/* audio compression */
-	CAudioCompression	AudioCompression;
+	CAudioCompression	AudioCompressionIn;
-	int					iAudComprSize;
+	int					iAudComprSizeIn;
+	CAudioCompression	AudioCompressionOut;
+	int					iAudComprSizeOut;
 
 	/* resampling */
 	CResample			ResampleObj;
@@ -125,6 +127,9 @@ protected:
 	int					iTimeStampActCnt;
 
 	int					iConTimeOut;
+	int					iConTimeOutStartVal;
+
+	int					iCurNetwInBlSiFact;
 
 	QMutex Mutex;
 

src/global.h

@@ -66,10 +66,14 @@
 
 /* first tests showed that with 24000 kHz a block time shorter than 5 ms leads to
    much higher DSL network latencies. A length of 6 ms seems to be optimal */
+#define NET_BLOCK_SIZE_FACTOR			3 // 3 * 2 ms = 6 ms
+
+// maximum value of factor for network block size
+#define NET_BLOCK_SIZE_FACTOR_MAX		10
+
 #define BLOCK_DURATION_MS				6 /* ms */
 
 #define BLOCK_SIZE_SAMPLES				( BLOCK_DURATION_MS * SAMPLE_RATE / 1000 )
-#define SND_CRD_BLOCK_SIZE_SAMPLES		( BLOCK_DURATION_MS * SND_CRD_SAMPLE_RATE / 1000 )
 
 /* maximum network buffer size (which can be chosen by slider) */
 #define MAX_NET_BUF_SIZE_NUM_BL			12 /* number of blocks */
@@ -97,7 +101,7 @@
 
 /* length of the moving average buffer for response time measurement */
 #define TIME_MOV_AV_RESPONSE			30 /* seconds */
-#define LEN_MOV_AV_RESPONSE				(TIME_MOV_AV_RESPONSE * 1000 / BLOCK_DURATION_MS)
+#define LEN_MOV_AV_RESPONSE				(TIME_MOV_AV_RESPONSE * 1000 / MIN_BLOCK_DURATION_MS)
 
 
 #define _MAXSHORT						32767

src/socket.h

@@ -38,7 +38,7 @@
 
 /* Definitions ****************************************************************/
 /* maximum block size for network input buffer. Consider two bytes per sample */
-#define MAX_SIZE_BYTES_NETW_BUF			(BLOCK_SIZE_SAMPLES * 2)
+#define MAX_SIZE_BYTES_NETW_BUF			( NET_BLOCK_SIZE_FACTOR_MAX * MIN_BLOCK_SIZE_SAMPLES * 2 )
 
 
 /* Classes ********************************************************************/

windows/sound.h

@@ -42,11 +42,8 @@
 #define MAX_SND_BUF_IN			200
 #define MAX_SND_BUF_OUT			200
 
-#define NUM_SOUND_BUFFERS_IN	(70 / BLOCK_DURATION_MS)
+#define NUM_SOUND_BUFFERS_IN	(70 / MIN_BLOCK_DURATION_MS)
-#define NUM_SOUND_BUFFERS_OUT	(80 / BLOCK_DURATION_MS)
+#define NUM_SOUND_BUFFERS_OUT	(80 / MIN_BLOCK_DURATION_MS)
-
-//#define NUM_SOUND_BUFFERS_IN	7//200		/* Number of sound card buffers */
-//#define NUM_SOUND_BUFFERS_OUT	8//100//15		/* Number of sound card buffers */
 
 /* Maximum number of recognized sound cards installed in the system */
 #define MAX_NUMBER_SOUND_CARDS	10