Rawaudio dev

ChangeLog

@@ -1,5 +1,7 @@
 ### 3.12.0dev <- NOTE: the release version number will be 4.0.0 ###
 
+- Add uncompressed audio transmission - dedicated to the memory of Hans Petter Selasky (1982 - 2023)
+  (contributed by @dingodoppelt)
 
 ### 3.12.0 (2026-05-02) ###
 

src/channel.cpp

@@ -83,6 +83,8 @@ CChannel::CChannel ( const bool bNIsServer ) :
 
     QObject::connect ( &Protocol, &CProtocol::ClientIDReceived, this, &CChannel::ClientIDReceived );
 
+    QObject::connect ( &Protocol, &CProtocol::RawAudioSupported, this, &CChannel::RawAudioSupported );
+
     QObject::connect ( &Protocol, &CProtocol::MuteStateHasChangedReceived, this, &CChannel::MuteStateHasChangedReceived );
 
     QObject::connect ( &Protocol, &CProtocol::ChangeChanInfo, this, &CChannel::OnChangeChanInfo );

src/channel.h

@@ -149,6 +149,7 @@ class CChannel : public QObject
         }
     }
     void CreateClientIDMes ( const int iChanID ) { Protocol.CreateClientIDMes ( iChanID ); }
+    void CreateRawAudioSupportedMes() { Protocol.CreateRawAudioSupportedMes(); }
     void CreateReqNetwTranspPropsMes() { Protocol.CreateReqNetwTranspPropsMes(); }
     void CreateReqSplitMessSupportMes() { Protocol.CreateReqSplitMessSupportMes(); }
     void CreateReqJitBufMes() { Protocol.CreateReqJitBufMes(); }
@@ -272,6 +273,7 @@ public slots:
     void ConClientListMesReceived ( CVector<CChannelInfo> vecChanInfo );
     void ChanInfoHasChanged();
     void ClientIDReceived ( int iChanID );
+    void RawAudioSupported();
     void MuteStateHasChanged ( int iChanID, bool bIsMuted );
     void MuteStateHasChangedReceived ( int iChanID, bool bIsMuted );
     void ReqChanInfo();

src/client.cpp

@@ -70,7 +70,8 @@ CClient::CClient ( const quint16  iPortNumber,
     bJitterBufferOK ( true ),
     bEnableIPv6 ( bNEnableIPv6 ),
     bMuteMeInPersonalMix ( bNMuteMeInPersonalMix ),
-    iServerSockBufNumFrames ( DEF_NET_BUF_SIZE_NUM_BL )
+    iServerSockBufNumFrames ( DEF_NET_BUF_SIZE_NUM_BL ),
+    bRawAudioIsSupported ( false )
 {
     int iOpusError;
 
@@ -131,6 +132,8 @@ CClient::CClient ( const quint16  iPortNumber,
 
     QObject::connect ( &Channel, &CChannel::ClientIDReceived, this, &CClient::OnClientIDReceived );
 
+    QObject::connect ( &Channel, &CChannel::RawAudioSupported, this, &CClient::OnRawAudioSupported );
+
     QObject::connect ( &Channel, &CChannel::MuteStateHasChangedReceived, this, &CClient::OnMuteStateHasChangedReceived );
 
     QObject::connect ( &Channel, &CChannel::LicenceRequired, this, &CClient::LicenceRequired );
@@ -989,6 +992,27 @@ void CClient::OnClientIDReceived ( int iServerChanID )
     emit ClientIDReceived ( iChanID );
 }
 
+void CClient::OnRawAudioSupported()
+{
+    if ( !bRawAudioIsSupported )
+    {
+        const bool bWasRunning = Sound.IsRunning();
+
+        if ( bWasRunning )
+        {
+            Sound.Stop();
+        }
+
+        bRawAudioIsSupported = true;
+        Init();
+
+        if ( bWasRunning )
+        {
+            Sound.Start();
+        }
+    }
+}
+
 void CClient::Start()
 {
     // init object
@@ -1017,6 +1041,10 @@ void CClient::Stop()
     // disable channel
     Channel.SetEnable ( false );
 
+    // Fall back to opus in case raw was used
+    bRawAudioIsSupported = false;
+    Init();
+
     // wait for approx. 100 ms to make sure no audio packet is still in the
     // network queue causing the channel to be reconnected right after having
     // received the disconnect message (seems not to gain much, disconnect is
@@ -1156,6 +1184,21 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported )
+                {
+                    // no OPUS encoding or decoding
+                    CurOpusEncoder = nullptr;
+                    CurOpusDecoder = nullptr;
+
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    // fall back to highest OPUS quality
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE;
+                }
+                break;
             }
         }
         else
@@ -1175,6 +1218,21 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported )
+                {
+                    // no OPUS encoding or decoding
+                    CurOpusEncoder = nullptr;
+                    CurOpusDecoder = nullptr;
+
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    // fall back to highest OPUS quality
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE;
+                }
+                break;
             }
         }
     }
@@ -1199,6 +1257,21 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported )
+                {
+                    // no OPUS encoding or decoding
+                    CurOpusEncoder = nullptr;
+                    CurOpusDecoder = nullptr;
+
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    // fall back to highest OPUS quality
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY;
+                }
+                break;
             }
         }
         else
@@ -1218,6 +1291,21 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported )
+                {
+                    // no OPUS encoding or decoding
+                    CurOpusEncoder = nullptr;
+                    CurOpusDecoder = nullptr;
+
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    // fall back to highest OPUS quality
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY;
+                }
+                break;
             }
         }
     }
@@ -1229,8 +1317,12 @@ void CClient::Init()
     vecZeros.Init ( iStereoBlockSizeSam, 0 );
     vecsStereoSndCrdMuteStream.Init ( iStereoBlockSizeSam );
 
-    opus_custom_encoder_ctl ( CurOpusEncoder,
-                              OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes, iOPUSFrameSizeSamples ) ) );
+    // In case we are connected to a non raw audio server or we don't use raw audio we need to initialze the codec
+    if ( CurOpusEncoder != nullptr )
+    {
+        opus_custom_encoder_ctl ( CurOpusEncoder,
+                                  OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes, iOPUSFrameSizeSamples ) ) );
+    }
 
     // inits for network and channel
     vecbyNetwData.Init ( iCeltNumCodedBytes );
@@ -1391,9 +1483,10 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
 
     for ( i = 0, j = 0; i < iSndCrdFrameSizeFactor; i++, j += iNumAudioChannels * iOPUSFrameSizeSamples )
     {
-        // OPUS encoding
+        // OPUS encoding or copying RAW audio?
         if ( CurOpusEncoder != nullptr )
         {
+            // OPUS encoding
             if ( bMuteOutStream )
             {
                 iUnused = opus_custom_encode ( CurOpusEncoder, &vecZeros[j], iOPUSFrameSizeSamples, &vecCeltData[0], iCeltNumCodedBytes );
@@ -1403,6 +1496,20 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
                 iUnused = opus_custom_encode ( CurOpusEncoder, &vecsStereoSndCrd[j], iOPUSFrameSizeSamples, &vecCeltData[0], iCeltNumCodedBytes );
             }
         }
+        else if ( bRawAudioIsSupported )
+        {
+            // RAW audio
+            if ( bMuteOutStream )
+            {
+                // output muted - fill with silence
+                memset ( &vecCeltData[0], 0, iCeltNumCodedBytes );
+            }
+            else
+            {
+                // copy raw audio data
+                memcpy ( &vecCeltData[0], &vecsStereoSndCrd[j], iCeltNumCodedBytes );
+            }
+        }
 
         // send coded audio through the network
         Channel.PrepAndSendPacket ( &Socket, vecCeltData, iCeltNumCodedBytes );
@@ -1437,11 +1544,26 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
             bJitterBufferOK = false;
         }
 
-        // OPUS decoding
+        // OPUS decoding or copying RAW audio?
         if ( CurOpusDecoder != nullptr )
         {
+            // OPUS decoding
             iUnused = opus_custom_decode ( CurOpusDecoder, pCurCodedData, iCeltNumCodedBytes, &vecsStereoSndCrd[j], iOPUSFrameSizeSamples );
         }
+        else if ( bRawAudioIsSupported )
+        {
+            // RAW audio
+            if ( pCurCodedData != nullptr )
+            {
+                // copy raw audio data
+                memcpy ( &vecsStereoSndCrd[j], pCurCodedData, iCeltNumCodedBytes );
+            }
+            else
+            {
+                // missing audio - fill with silence
+                memset ( &vecsStereoSndCrd[j], 0, iCeltNumCodedBytes );
+            }
+        }
     }
 
     // for muted stream we have to add our local data here
@@ -1488,7 +1610,7 @@ int CClient::EstimatedOverallDelay ( const int iPingTimeMs )
     // length. Since that is usually not the case but the buffers are usually
     // a bit larger than necessary, we introduce some factor for compensation.
     // Consider the jitter buffer on the client and on the server side, too.
-    const float fTotalJitterBufferDelayMs = fSystemBlockDurationMs * ( GetSockBufNumFrames() + GetServerSockBufNumFrames() ) * 0.7f;
+    const float fTotalJitterBufferDelayMs = fSystemBlockDurationMs * ( GetSockBufNumFrames() + GetServerSockBufNumFrames() ) * JITTBUF_COMP_FACTOR;
 
     // consider delay introduced by the sound card conversion buffer by using
     // "GetSndCrdConvBufAdditionalDelayMonoBlSize()"
@@ -1519,7 +1641,7 @@ int CClient::EstimatedOverallDelay ( const int iPingTimeMs )
     const float fDelayToFillNetworkPacketsMs = GetSystemMonoBlSize() * 1000.0f / SYSTEM_SAMPLE_RATE_HZ;
 
     // OPUS additional delay at small frame sizes is half a frame size
-    const float fAdditionalAudioCodecDelayMs = fSystemBlockDurationMs / 2;
+    const float fAdditionalAudioCodecDelayMs = CurOpusDecoder != nullptr ? fSystemBlockDurationMs / 2 : 0.0f;
 
     const float fTotalBufferDelayMs =
         fDelayToFillNetworkPacketsMs + fTotalJitterBufferDelayMs + fTotalSoundCardDelayMs + fAdditionalAudioCodecDelayMs;

src/client.h

@@ -410,7 +410,8 @@ class CClient : public QObject
     QMutex MutexDriverReinit;
 
     // server settings
-    int iServerSockBufNumFrames;
+    int  iServerSockBufNumFrames;
+    bool bRawAudioIsSupported;
 
     // for ping measurement
     QElapsedTimer PreciseTime;
@@ -453,6 +454,7 @@ protected slots:
     void OnControllerInFaderIsMute ( int iChannelIdx, bool bIsMute );
     void OnControllerInMuteMyself ( bool bMute );
     void OnClientIDReceived ( int iServerChanID );
+    void OnRawAudioSupported();
     void OnMuteStateHasChangedReceived ( int iServerChanID, bool bIsMuted );
     void OnCLChannelLevelListReceived ( CHostAddress InetAddr, CVector<uint16_t> vecLevelList );
     void OnConClientListMesReceived ( CVector<CChannelInfo> vecChanInfo );

src/clientsettingsdlg.cpp

@@ -494,6 +494,7 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, CClientSettings* pNSet
     cbxAudioQuality->addItem ( tr ( "Low" ) );    // AQ_LOW
     cbxAudioQuality->addItem ( tr ( "Normal" ) ); // AQ_NORMAL
     cbxAudioQuality->addItem ( tr ( "High" ) );   // AQ_HIGH
+    cbxAudioQuality->addItem ( tr ( "Max" ) );    // AQ_RAW
     cbxAudioQuality->setCurrentIndex ( static_cast<int> ( pClient->GetAudioQuality() ) );
 
     // GUI design (skin) combo box

src/global.h

@@ -230,6 +230,9 @@ LED bar:      lbr
 // defines the time interval at which the ping time is updated in the GUI
 #define PING_UPDATE_TIME_MS 500 // ms
 
+// defines a factor to compensate for larger than ideal jitter buffer sizes for estimated overall delay calculation
+#define JITTBUF_COMP_FACTOR 0.7f
+
 // defines the time interval at which the ping time is updated for the server list
 #define PING_UPDATE_TIME_SERVER_LIST_MS 2500 // ms
 

src/protocol.cpp

@@ -235,6 +235,11 @@ MESSAGES (with connection)
     note: does not have any data -> n = 0
 
 
+- PROTMESSID_RAWAUDIO_SUPPORTED: informs client that server supports raw (uncompressed) audio
+
+    note: does not have any data -> n = 0
+
+
 - PROTMESSID_RECORDER_STATE: notifies of changes in the server jam recorder state
 
     +--------------+
@@ -813,6 +818,10 @@ void CProtocol::ParseMessageBody ( const CVector<uint8_t>& vecbyMesBodyData, con
                     EvaluateSplitMessSupportedMes();
                     break;
 
+                case PROTMESSID_RAWAUDIO_SUPPORTED:
+                    EvaluateRawAudioSupportedMes();
+                    break;
+
                 case PROTMESSID_LICENCE_REQUIRED:
                     EvaluateLicenceRequiredMes ( vecbyMesBodyDataRef );
                     break;
@@ -1520,6 +1529,16 @@ bool CProtocol::EvaluateSplitMessSupportedMes()
     return false; // no error
 }
 
+void CProtocol::CreateRawAudioSupportedMes() { CreateAndSendMessage ( PROTMESSID_RAWAUDIO_SUPPORTED, CVector<uint8_t> ( 0 ) ); }
+
+bool CProtocol::EvaluateRawAudioSupportedMes()
+{
+    // invoke message action
+    emit RawAudioSupported();
+
+    return false; // no error
+}
+
 void CProtocol::CreateLicenceRequiredMes ( const ELicenceType eLicenceType )
 {
     CVector<uint8_t> vecData ( 1 ); // 1 bytes of data

src/protocol.h

@@ -62,6 +62,7 @@
 #define PROTMESSID_RECORDER_STATE           33 // contains the state of the jam recorder (ERecorderState)
 #define PROTMESSID_REQ_SPLIT_MESS_SUPPORT   34 // request support for split messages
 #define PROTMESSID_SPLIT_MESS_SUPPORTED     35 // split messages are supported
+#define PROTMESSID_RAWAUDIO_SUPPORTED       36 // raw (uncompressed) audio is supported
 
 // message IDs of connection less messages (CLM)
 // DEFINITION -> start at 1000, end at 1999, see IsConnectionLessMessageID
@@ -124,6 +125,7 @@ class CProtocol : public QObject
     void CreateReqNetwTranspPropsMes();
     void CreateReqSplitMessSupportMes();
     void CreateSplitMessSupportedMes();
+    void CreateRawAudioSupportedMes();
     void CreateLicenceRequiredMes ( const ELicenceType eLicenceType );
     void CreateOpusSupportedMes();
 
@@ -258,6 +260,7 @@ class CProtocol : public QObject
     bool EvaluateReqNetwTranspPropsMes();
     bool EvaluateReqSplitMessSupportMes();
     bool EvaluateSplitMessSupportedMes();
+    bool EvaluateRawAudioSupportedMes();
     bool EvaluateLicenceRequiredMes ( const CVector<uint8_t>& vecData );
     bool EvaluateVersionAndOSMes ( const CVector<uint8_t>& vecData );
     bool EvaluateRecorderStateMes ( const CVector<uint8_t>& vecData );
@@ -321,6 +324,7 @@ public slots:
     void ReqNetTranspProps();
     void ReqSplitMessSupport();
     void SplitMessSupported();
+    void RawAudioSupported();
     void LicenceRequired ( ELicenceType eLicenceType );
     void VersionAndOSReceived ( COSUtil::EOpSystemType eOSType, QString strVersion );
     void RecorderStateReceived ( ERecorderState eRecorderState );