Allow the option to select sample rate to stream at rather than have it hard coded to 48000hz

NoiseGate/NoiseGate.cpp

@@ -74,8 +74,7 @@ void NoiseGateFilter::ApplyNoiseGate(float *buffer, int totalFloats)
     if(totalFloats % 2)
         return; // Odd number of samples
 
-    // OBS is currently hard-coded to 48ksps
+    const float SAMPLE_RATE_F = float(OBSGetSampleRateHz());
-    const float SAMPLE_RATE_F = 48000.0f;
     const float dtPerSample = 1.0f / SAMPLE_RATE_F;
 
     // Convert configuration times into per-sample amounts
@@ -83,10 +82,9 @@ void NoiseGateFilter::ApplyNoiseGate(float *buffer, int totalFloats)
     const float releaseRate = 1.0f / (parent->releaseTime * SAMPLE_RATE_F);
 
     // Determine level decay rate. We don't want human voice (75-300Hz) to cross the close
-    // threshold if the previous peak crosses the open threshold. 75Hz at 48ksps is 640
+    // threshold if the previous peak crosses the open threshold.
-    // samples between peaks.
     const float thresholdDiff = parent->openThreshold - parent->closeThreshold;
-    const float minDecayPeriod = (1.0f / 75.0f) * SAMPLE_RATE_F; // 640 samples
+    const float minDecayPeriod = (1.0f / 75.0f) * SAMPLE_RATE_F;
     const float decayRate = thresholdDiff / minDecayPeriod;
 
     // We can't use SSE as the processing of each sample depends on the processed

OBS.rc

@@ -126,7 +126,7 @@ BEGIN
     RTEXT           "Settings.Encoding.Audio.Bitrate",IDC_STATIC,221,85,128,8
     COMBOBOX        IDC_AUDIOBITRATE,354,83,61,62,CBS_DROPDOWNLIST | WS_VSCROLL | WS_TABSTOP
     RTEXT           "Settings.Encoding.Audio.Format",IDC_STATIC,10,101,138,8
-    COMBOBOX        IDC_AUDIOFORMAT,152,99,264,62,CBS_DROPDOWNLIST | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
+    COMBOBOX        IDC_AUDIOFORMAT,152,99,264,62,CBS_DROPDOWNLIST | WS_VSCROLL | WS_TABSTOP
     LTEXT           "Settings.Info",IDC_INFO,4,122,417,37,NOT WS_VISIBLE
 END
 

OBSApi/APIDefs.cpp

@@ -141,4 +141,6 @@ void OBSGetCurMicVolumeStats(float *rms, float *max, float *peak)       {API->Ge
 void OBSAddSettingsPane(SettingsPane *pane)     {API->AddSettingsPane(pane);}
 void OBSRemoveSettingsPane(SettingsPane *pane)  {API->RemoveSettingsPane(pane);}
 
-UINT OBSGetAPIVersion()                         {return 0x0100;}
+UINT OBSGetAPIVersion()                         {return 0x0101;}
+
+UINT OBSGetSampleRateHz()                       {return API->GetSampleRateHz();}

OBSApi/APIInterface.h

@@ -172,6 +172,8 @@ public:
     virtual Vect2 MapFrameToWindowSize(Vect2 frameSize) const=0;
     virtual Vect2 GetWindowToFrameScale() const=0;
     virtual Vect2 GetFrameToWindowScale() const=0;
+
+    virtual UINT GetSampleRateHz() const=0;
 };
 
 BASE_EXPORT extern APIInterface *API;
@@ -286,3 +288,5 @@ BASE_EXPORT void OBSRemoveSettingsPane(SettingsPane *pane);
 
 /** gets API version.  version is formatted: 0xMMmm */
 BASE_EXPORT UINT OBSGetAPIVersion();
+
+BASE_EXPORT UINT OBSGetSampleRateHz();

OBSApi/AudioSource.cpp

@@ -87,7 +87,9 @@ void AudioSource::InitAudioData(bool bFloat, UINT channels, UINT samplesPerSec,
 
     //-----------------------------
 
-    if(inputSamplesPerSec != 48000)
+    UINT sampleRateHz = OBSGetSampleRateHz();
+
+    if(inputSamplesPerSec != sampleRateHz)
     {
         int errVal;
 
@@ -96,12 +98,11 @@ void AudioSource::InitAudioData(bool bFloat, UINT channels, UINT samplesPerSec,
         if(!resampler)
             CrashError(TEXT("AudioSource::InitAudioData: Could not initiate resampler"));
 
-        resampleRatio = 48000.0 / double(inputSamplesPerSec);
+        resampleRatio = double(sampleRateHz) / double(inputSamplesPerSec);
         bResample = true;
 
         //----------------------------------------------------
         // hack to get rid of that weird first quirky resampled packet size
-        // (always returns a non-480 sized packet on the first resample)
 
         SRC_DATA data;
         data.src_ratio = resampleRatio;
@@ -679,7 +680,7 @@ bool AudioSource::GetBuffer(float **buffer, QWORD targetTimestamp)
         }
     }
 
-    outputBuffer.SetSize(480*2);
+    outputBuffer.SetSize(OBSGetSampleRateHz()/100*2);
 
     *buffer = outputBuffer.Array();
 

Source/API.cpp

@@ -539,6 +539,8 @@ public:
 
     virtual void AddSettingsPane(SettingsPane *pane)    {App->AddSettingsPane(pane);}
     virtual void RemoveSettingsPane(SettingsPane *pane) {App->RemoveSettingsPane(pane);}
+
+    virtual UINT GetSampleRateHz() const {return App->GetSampleRateHz();}
 };
 
 APIInterface* CreateOBSApiInterface()

Source/Encoder_AAC.cpp

@@ -46,7 +46,7 @@ public:
     {
         curBitRate = bitRate;
 
-        faac = faacEncOpen(48000, 2, &numReadSamples, &outputSize);
+        faac = faacEncOpen(App->GetSampleRateHz(), 2, &numReadSamples, &outputSize);
 
         //Log(TEXT("numReadSamples: %d"), numReadSamples);
         aacBuffer.SetSize(outputSize+2);
@@ -155,7 +155,7 @@ public:
             inputBuffer.RemoveRange(0, numReadSamples);
 
             bufferedTimestamps << curEncodeTimestamp;
-            curEncodeTimestamp = curTimestamp + (((numReadSamples-lastSampleSize)/2)/48);
+            curEncodeTimestamp = curTimestamp + (((numReadSamples-lastSampleSize)/2)*1000/App->GetSampleRateHz());
         }
 
         return ret > 0;

Source/Encoder_MP3.cpp

@@ -59,8 +59,8 @@ public:
         if(!lgf)
             CrashError(TEXT("Unable to open mp3 encoder"));
 
-        lame_set_in_samplerate(lgf, 48000);
+        lame_set_in_samplerate(lgf, App->GetSampleRateHz());
-        lame_set_out_samplerate(lgf, 48000);
+        lame_set_out_samplerate(lgf, App->GetSampleRateHz());
         lame_set_num_channels(lgf, 2);
         lame_set_mode(lgf, STEREO);
         lame_set_disable_reservoir(lgf, TRUE); //bit reservoir has to be disabled for seamless streaming
@@ -102,7 +102,7 @@ public:
             frameCounter -= outputFrameSize;
 
             bufferedTimestamps << curEncodeTimestamp;
-            curEncodeTimestamp = timestamp + ((outputFrameSize-lastSampleSize)/48);
+            curEncodeTimestamp = timestamp + ((outputFrameSize-lastSampleSize)*1000/App->GetSampleRateHz());
         }
 
         int ret = lame_encode_buffer_interleaved_ieee_float(lgf, (float*)input, numInputFrames, MP3OutputBuffer.Array()+1, dwMP3MaxSize);

Source/MP4FileStream.cpp

@@ -57,7 +57,7 @@ struct MP4AudioFrameInfo
 
 inline UINT64 ConvertToAudioTime(DWORD timestamp, UINT64 minVal)
 {
-    UINT val = UINT64(timestamp)*48000/1000;
+    UINT val = UINT64(timestamp)*App->GetSampleRateHz()/1000;
     return MAX(val, minVal);
 }
 
@@ -303,7 +303,7 @@ public:
 
         DWORD macTime = fastHtonl(DWORD(GetMacTime()));
         UINT videoDuration = fastHtonl(lastVideoTimestamp + App->GetFrameTime());
-        UINT audioDuration = fastHtonl(lastVideoTimestamp + DWORD(double(audioFrameSize)/48));
+        UINT audioDuration = fastHtonl(lastVideoTimestamp + DWORD(double(audioFrameSize)*1000.0/double(App->GetSampleRateHz())));
         UINT width, height;
         App->GetOutputSize(width, height);
 
@@ -453,7 +453,7 @@ public:
                 output.OutputDword(0); //version and flags (none)
                 output.OutputDword(macTime); //creation time
                 output.OutputDword(macTime); //modified time
-                output.OutputDword(DWORD_BE(48000)); //time scale
+                output.OutputDword(DWORD_BE(App->GetSampleRateHz())); //time scale
                 output.OutputDword(audioUnitDuration);
                 output.OutputDword(bMP3 ? DWORD_BE(0x55c40000) : DWORD_BE(0x15c70000));
               PopBox(output); //mdhd
@@ -495,7 +495,7 @@ public:
                       output.OutputWord(WORD_BE(16)); //sample size
                       output.OutputWord(0); //quicktime audio compression id
                       output.OutputWord(0); //quicktime audio packet size
-                      output.OutputDword(DWORD_BE(48000<<16)); //sample rate (fixed point)
+                      output.OutputDword(DWORD_BE(App->GetSampleRateHz()<<16)); //sample rate (fixed point)
                       PushBox(output, DWORD_BE('esds'));
                         output.OutputDword(0); //version and flags (none)
                         output.OutputByte(3); //ES descriptor type

Source/OBS.h

@@ -499,6 +499,8 @@ class OBS
     AudioSource  *micAudio;
     List<AudioSource*> auxAudioSources;
 
+    UINT sampleRateHz;
+
     AudioEncoder *audioEncoder;
 
     //---------------------------------------------------
@@ -913,6 +915,8 @@ public:
         OSLeaveMutex(hAuxAudioMutex);
     }
 
+    inline UINT GetSampleRateHz() const {return sampleRateHz;}
+
     inline QWORD GetAudioTime() const {return latestAudioTime;}
     inline QWORD GetVideoTime() const {return latestVideoTime;}
 

Source/OBSCapture.cpp

@@ -284,7 +284,20 @@ retryHookTestV2:
         }
     }
 
-    //-------------------------------------------------------------
+    //------------------------------------------------------------------
+
+    UINT format = AppConfig->GetInt(L"Audio Encoding", L"Format", 1);
+
+    switch (format) {
+    case 0: sampleRateHz = 44100; break;
+    default:
+    case 1: sampleRateHz = 48000; break;
+    }
+
+    Log(L"------------------------------------------");
+    Log(L"Audio Format: %uhz", sampleRateHz);
+
+    //------------------------------------------------------------------
 
     AudioDeviceList playbackDevices;
     GetAudioDevices(playbackDevices, ADT_PLAYBACK);
@@ -881,11 +894,11 @@ void OBS::EncodeAudioSegment(float *buffer, UINT numFrames, QWORD timestamp)
     }
 }
 
-const int audioSamplesPerSec = 48000;
-const int audioSampleSize = audioSamplesPerSec/100;
-
 void OBS::MainAudioLoop()
 {
+    const unsigned int audioSamplesPerSec = App->GetSampleRateHz();
+    const unsigned int audioSampleSize = audioSamplesPerSec/100;
+
     DWORD taskID = 0;
     HANDLE hTask = AvSetMmThreadCharacteristics(TEXT("Pro Audio"), &taskID);
 

Source/RTMPStuff.cpp

@@ -265,7 +265,7 @@ char* OBS::EncMetaData(char *enc, char *pend, bool bFLVFile)
         enc = AMF_EncodeNamedString(enc, pend, &av_audiocodecid,    av_codecFourCC);//audioCodecID);//
 
     enc = AMF_EncodeNamedNumber(enc, pend, &av_audiodatarate,   double(audioBitRate)); //ex. 128kb\s
-    enc = AMF_EncodeNamedNumber(enc, pend, &av_audiosamplerate, 48000.0);
+    enc = AMF_EncodeNamedNumber(enc, pend, &av_audiosamplerate, double(App->GetSampleRateHz()));
     enc = AMF_EncodeNamedNumber(enc, pend, &av_audiosamplesize, 16.0);
     enc = AMF_EncodeNamedNumber(enc, pend, &av_audiochannels,   2.0);
     enc = AMF_EncodeNamedBoolean(enc, pend, &av_stereo,         true);

Source/SettingsEncoding.cpp

@@ -213,7 +213,8 @@ INT_PTR SettingsEncoding::ProcMessage(UINT message, WPARAM wParam, LPARAM lParam
                 //--------------------------------------------
 
                 hwndTemp = GetDlgItem(hwnd, IDC_AUDIOFORMAT);
-                SendMessage(hwndTemp, CB_ADDSTRING, 0, (LPARAM)TEXT("48khz mono"));
+                //SendMessage(hwndTemp, CB_ADDSTRING, 0, (LPARAM)TEXT("48khz mono"));
+                SendMessage(hwndTemp, CB_ADDSTRING, 0, (LPARAM)TEXT("44.1khz stereo"));
                 SendMessage(hwndTemp, CB_ADDSTRING, 0, (LPARAM)TEXT("48khz stereo"));
 
                 LoadSettingComboInt(hwndTemp, TEXT("Audio Encoding"), TEXT("Format"), 1, 1);