Add 16-bit PCM uncompressed audio transmission

Author: dingodoppelt

ChangeLog

@@ -5,6 +5,8 @@
 - Client: Fix occasional spurious version updated message display (#3691).
   (contributed by @softins)
 
+- Add uncompressed audio transmission - dedicated to the memory of Hans Petter Selasky (1982 - 2023)
+  (contributed by @dingodoppelt)
 
 ### 3.12.0 (2026-05-02) ###
 

src/client.cpp

@@ -70,7 +70,9 @@ 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 ),
+    bUseRawAudio ( false )
 {
     int iOpusError;
 
@@ -135,7 +137,7 @@ CClient::CClient ( const quint16  iPortNumber,
 
     QObject::connect ( &Channel, &CChannel::LicenceRequired, this, &CClient::LicenceRequired );
 
-    QObject::connect ( &Channel, &CChannel::VersionAndOSReceived, this, &CClient::VersionAndOSReceived );
+    QObject::connect ( &Channel, &CChannel::VersionAndOSReceived, this, &CClient::OnVersionAndOSReceived );
 
     QObject::connect ( &Channel, &CChannel::RecorderStateReceived, this, &CClient::RecorderStateReceived );
 
@@ -395,6 +397,32 @@ void CClient::OnConClientListMesReceived ( CVector<CChannelInfo> vecChanInfo )
     emit ConClientListMesReceived ( vecChanInfo );
 }
 
+void CClient::OnVersionAndOSReceived ( COSUtil::EOpSystemType eOSType, QString strVersion )
+{
+#if QT_VERSION >= QT_VERSION_CHECK( 5, 6, 0 )
+    const bool bWasRunning = Sound.IsRunning();
+    if ( bWasRunning )
+    {
+        Sound.Stop();
+    }
+    if ( QVersionNumber::compare ( QVersionNumber::fromString ( strVersion ), QVersionNumber ( 3, 11, 1 ) ) == 0 )
+    {
+        bRawAudioIsSupported = true;
+        Init();
+    }
+    else
+    {
+        bRawAudioIsSupported = false;
+        Init();
+    }
+    if ( bWasRunning )
+    {
+        Sound.Start();
+    }
+#endif
+    emit VersionAndOSReceived ( eOSType, strVersion );
+}
+
 void CClient::CreateServerJitterBufferMessage()
 {
     // per definition in the client: if auto jitter buffer is enabled, both,
@@ -1017,6 +1045,11 @@ void CClient::Stop()
     // disable channel
     Channel.SetEnable ( false );
 
+    // Fall back to opus in case raw was used
+    bRawAudioIsSupported = false;
+    bUseRawAudio         = 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 +1189,16 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported && Channel.IsEnabled() )
+                {
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE;
+                }
+                break;
             }
         }
         else
@@ -1175,6 +1218,16 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported && Channel.IsEnabled() )
+                {
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE;
+                }
+                break;
             }
         }
     }
@@ -1199,6 +1252,16 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported && Channel.IsEnabled() )
+                {
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY;
+                }
+                break;
             }
         }
         else
@@ -1218,19 +1281,36 @@ void CClient::Init()
             case AQ_HIGH:
                 iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY;
                 break;
+            case AQ_RAW:
+                if ( bRawAudioIsSupported && Channel.IsEnabled() )
+                {
+                    iCeltNumCodedBytes = sizeof ( int16_t ) * iNumAudioChannels * iOPUSFrameSizeSamples;
+                }
+                else
+                {
+                    iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY;
+                }
+                break;
             }
         }
     }
 
+    // determine whether to use raw audio
+    bUseRawAudio = bRawAudioIsSupported && eAudioQuality == AQ_RAW;
+
     // calculate stereo (two channels) buffer size
     iStereoBlockSizeSam = 2 * iMonoBlockSizeSam;
 
     vecCeltData.Init ( iCeltNumCodedBytes );
     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 ( !bUseRawAudio )
+    {
+        opus_custom_encoder_ctl ( CurOpusEncoder,
+                                  OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes, iOPUSFrameSizeSamples ) ) );
+    }
 
     // inits for network and channel
     vecbyNetwData.Init ( iCeltNumCodedBytes );
@@ -1391,19 +1471,33 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
 
     for ( i = 0, j = 0; i < iSndCrdFrameSizeFactor; i++, j += iNumAudioChannels * iOPUSFrameSizeSamples )
     {
-        // OPUS encoding
-        if ( CurOpusEncoder != nullptr )
+        if ( !bUseRawAudio )
+        {
+            // OPUS encoding
+            if ( CurOpusEncoder != nullptr )
+            {
+                if ( bMuteOutStream )
+                {
+                    iUnused = opus_custom_encode ( CurOpusEncoder, &vecZeros[j], iOPUSFrameSizeSamples, &vecCeltData[0], iCeltNumCodedBytes );
+                }
+                else
+                {
+                    iUnused = opus_custom_encode ( CurOpusEncoder, &vecsStereoSndCrd[j], iOPUSFrameSizeSamples, &vecCeltData[0], iCeltNumCodedBytes );
+                }
+            }
+        }
+        else
         {
             if ( bMuteOutStream )
             {
-                iUnused = opus_custom_encode ( CurOpusEncoder, &vecZeros[j], iOPUSFrameSizeSamples, &vecCeltData[0], iCeltNumCodedBytes );
+                memset ( &vecCeltData[0], 0, iCeltNumCodedBytes );
             }
             else
             {
-                iUnused = opus_custom_encode ( CurOpusEncoder, &vecsStereoSndCrd[j], iOPUSFrameSizeSamples, &vecCeltData[0], iCeltNumCodedBytes );
+                // Send raw samples instead of OPUS
+                memcpy ( &vecCeltData[0], &vecsStereoSndCrd[j], iCeltNumCodedBytes );
             }
         }
-
         // send coded audio through the network
         Channel.PrepAndSendPacket ( &Socket, vecCeltData, iCeltNumCodedBytes );
     }
@@ -1423,23 +1517,36 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
         // get pointer to coded data and manage the flags
         if ( bReceiveDataOk )
         {
-            pCurCodedData = &vecbyNetwData[0];
+            if ( eAudioQuality == AQ_RAW && bRawAudioIsSupported )
+            {
+                memcpy ( &vecsStereoSndCrd[j], &vecbyNetwData[0], iCeltNumCodedBytes );
+                pCurCodedData = nullptr;
+            }
+            else
+            {
+                pCurCodedData = &vecbyNetwData[0];
+            }
 
             // on any valid received packet, we clear the initialization phase flag
             bIsInitializationPhase = false;
         }
         else
         {
+            if ( eAudioQuality == AQ_RAW && bRawAudioIsSupported )
+            {
+                memset ( &vecsStereoSndCrd[j], 0, iCeltNumCodedBytes );
+            }
+
             // for lost packets use null pointer as coded input data
             pCurCodedData = nullptr;
 
             // invalidate the buffer OK status flag
             bJitterBufferOK = false;
         }
 
-        // OPUS decoding
-        if ( CurOpusDecoder != nullptr )
+        if ( !bUseRawAudio && CurOpusDecoder != nullptr )
         {
+            // OPUS decoding
             iUnused = opus_custom_decode ( CurOpusDecoder, pCurCodedData, iCeltNumCodedBytes, &vecsStereoSndCrd[j], iOPUSFrameSizeSamples );
         }
     }
@@ -1488,7 +1595,7 @@ 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 +1626,7 @@ 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 = ( eAudioQuality == AQ_RAW && bRawAudioIsSupported ) ? 0.0f : fSystemBlockDurationMs / 2;
 
     const float fTotalBufferDelayMs =
         fDelayToFillNetworkPacketsMs + fTotalJitterBufferDelayMs + fTotalSoundCardDelayMs + fAdditionalAudioCodecDelayMs;

src/client.h

@@ -368,6 +368,7 @@ class CClient : public QObject
     bool                   bMuteOutStream;
     float                  fMuteOutStreamGain;
     CVector<unsigned char> vecCeltData;
+    bool                   bUseRawAudio;
 
     CHighPrioSocket         Socket;
     CSound                  Sound;
@@ -410,7 +411,8 @@ class CClient : public QObject
     QMutex MutexDriverReinit;
 
     // server settings
-    int iServerSockBufNumFrames;
+    int  iServerSockBufNumFrames;
+    bool bRawAudioIsSupported;
 
     // for ping measurement
     QElapsedTimer PreciseTime;
@@ -456,6 +458,7 @@ protected slots:
     void OnMuteStateHasChangedReceived ( int iServerChanID, bool bIsMuted );
     void OnCLChannelLevelListReceived ( CHostAddress InetAddr, CVector<uint16_t> vecLevelList );
     void OnConClientListMesReceived ( CVector<CChannelInfo> vecChanInfo );
+    void OnVersionAndOSReceived ( COSUtil::EOpSystemType eOSType, QString strVersion );
 
 signals:
     void ConClientListMesReceived ( 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