Remove automatic audio configuration permanently due to brokenness and allow renderers to request reinitialization on demand

app/cli/commandlineparser.cpp

@@ -194,7 +194,6 @@ StreamCommandLineParser::StreamCommandLineParser()
         {"borderless", StreamingPreferences::WM_FULLSCREEN_DESKTOP},
     };
     m_AudioConfigMap = {
-        {"auto",     StreamingPreferences::AC_AUTO},
         {"stereo",   StreamingPreferences::AC_FORCE_STEREO},
         {"surround", StreamingPreferences::AC_FORCE_SURROUND},
     };

app/gui/SettingsView.qml

@@ -406,10 +406,6 @@ ScrollView {
                     textRole: "text"
                     model: ListModel {
                         id: audioListModel
-                        ListElement {
-                            text: "Autodetect"
-                            val: StreamingPreferences.AC_AUTO
-                        }
                         ListElement {
                             text: "Stereo"
                             val: StreamingPreferences.AC_FORCE_STEREO

app/streaming/audio/audio.cpp

@@ -52,23 +52,6 @@ bool Session::testAudio(int audioConfiguration)
     return ret;
 }
 
-int Session::detectAudioConfiguration()
-{
-    IAudioRenderer* audioRenderer;
-
-    audioRenderer = createAudioRenderer();
-    if (audioRenderer == nullptr) {
-        // Hope for the best
-        return AUDIO_CONFIGURATION_STEREO;
-    }
-
-    int audioConfig = audioRenderer->detectAudioConfiguration();
-
-    delete audioRenderer;
-
-    return audioConfig;
-}
-
 int Session::arInit(int /* audioConfiguration */,
                     const POPUS_MULTISTREAM_CONFIGURATION opusConfig,
                     void* /* arContext */, int /* arFlags */)
@@ -92,15 +75,10 @@ int Session::arInit(int /* audioConfiguration */,
     }
 
     s_ActiveSession->m_AudioRenderer = s_ActiveSession->createAudioRenderer();
-    if (s_ActiveSession->m_AudioRenderer == nullptr) {
-        opus_multistream_decoder_destroy(s_ActiveSession->m_OpusDecoder);
-        return -2;
-    }
-
     if (!s_ActiveSession->m_AudioRenderer->prepareForPlayback(opusConfig)) {
         delete s_ActiveSession->m_AudioRenderer;
         opus_multistream_decoder_destroy(s_ActiveSession->m_OpusDecoder);
-        return -3;
+        return -2;
     }
 
     SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
@@ -123,16 +101,40 @@ void Session::arDecodeAndPlaySample(char* sampleData, int sampleLength)
 {
     int samplesDecoded;
 
-    samplesDecoded = opus_multistream_decode(s_ActiveSession->m_OpusDecoder,
+    s_ActiveSession->m_AudioSampleCount++;
-                                             (unsigned char*)sampleData,
+
-                                             sampleLength,
+    if (s_ActiveSession->m_AudioRenderer != nullptr) {
-                                             s_ActiveSession->m_OpusDecodeBuffer,
+        samplesDecoded = opus_multistream_decode(s_ActiveSession->m_OpusDecoder,
-                                             SAMPLES_PER_FRAME,
+                                                 (unsigned char*)sampleData,
-                                             0);
+                                                 sampleLength,
-    if (samplesDecoded > 0) {
+                                                 s_ActiveSession->m_OpusDecodeBuffer,
-        s_ActiveSession->m_AudioRenderer->submitAudio(s_ActiveSession->m_OpusDecodeBuffer,
+                                                 SAMPLES_PER_FRAME,
-                                                      static_cast<int>(sizeof(short) *
+                                                 0);
-                                                        samplesDecoded *
+        if (samplesDecoded > 0) {
-                                                        s_ActiveSession->m_AudioConfig.channelCount));
+            if (!s_ActiveSession->m_AudioRenderer->submitAudio(s_ActiveSession->m_OpusDecodeBuffer,
+                                                               static_cast<int>(
+                                                                   sizeof(short) *
+                                                                   samplesDecoded *
+                                                                   s_ActiveSession->m_AudioConfig.channelCount))) {
+
+                SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                            "Reinitializing audio renderer after failure");
+
+                delete s_ActiveSession->m_AudioRenderer;
+                s_ActiveSession->m_AudioRenderer = nullptr;
+            }
+        }
+    }
+
+    // Only try to recreate the audio renderer every 200 samples (1 second)
+    // to avoid thrashing if the audio device is unavailable. It is
+    // safe to reinitialize here because we can't be torn down while
+    // the audio decoder/playback thread is still alive.
+    if (s_ActiveSession->m_AudioRenderer == nullptr && (s_ActiveSession->m_AudioSampleCount % 200) == 0) {
+        s_ActiveSession->m_AudioRenderer = s_ActiveSession->createAudioRenderer();
+        if (!s_ActiveSession->m_AudioRenderer->prepareForPlayback(&s_ActiveSession->m_AudioConfig)) {
+            delete s_ActiveSession->m_AudioRenderer;
+            s_ActiveSession->m_AudioRenderer = nullptr;
+        }
     }
 }

app/streaming/audio/renderers/portaudiorenderer.cpp

@@ -74,14 +74,14 @@ bool PortAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
     return true;
 }
 
-void PortAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
+bool PortAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
 {
     SDL_assert(audioSize == SAMPLES_PER_FRAME * m_ChannelCount * 2);
 
     // Check if there is space for this sample in the buffer. Again, this can race
     // but in the worst case, we'll not see the sample callback having consumed a sample.
     if (((m_WriteIndex + 1) % CIRCULAR_BUFFER_SIZE) == m_ReadIndex) {
-        return;
+        return true;
     }
 
     SDL_memcpy(&m_AudioBuffer[m_WriteIndex * CIRCULAR_BUFFER_STRIDE], audioBuffer, audioSize);
@@ -102,33 +102,13 @@ void PortAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
             SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
                          "Pa_StartStream() failed: %s",
                          Pa_GetErrorText(error));
-            return;
+            return false;
         }
 
         m_Started = true;
     }
-}
 
-int PortAudioRenderer::detectAudioConfiguration() const
+    return true;
-{
-    const PaDeviceInfo* deviceInfo = Pa_GetDeviceInfo(Pa_GetDefaultOutputDevice());
-    if (deviceInfo == nullptr) {
-        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
-                     "Pa_GetDeviceInfo() failed");
-        return false;
-    }
-
-    // PulseAudio reports max output channels that don't
-    // correspond to any output devices (32 channels), so
-    // only use 5.1 surround sound if the output channel count
-    // is reasonable. Additionally, PortAudio doesn't do remixing
-    // for quadraphonic, so only use 5.1 if we have 6 or more channels.
-    if (deviceInfo->maxOutputChannels == 6 || deviceInfo->maxOutputChannels == 8) {
-        return AUDIO_CONFIGURATION_51_SURROUND;
-    }
-    else {
-        return AUDIO_CONFIGURATION_STEREO;
-    }
 }
 
 int PortAudioRenderer::paStreamCallback(const void*, void* output, unsigned long frameCount, const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags, void* userData)

app/streaming/audio/renderers/portaudiorenderer.h

@@ -18,9 +18,7 @@ public:
 
     virtual bool prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig);
 
-    virtual void submitAudio(short* audioBuffer, int audioSize);
+    virtual bool submitAudio(short* audioBuffer, int audioSize);
-
-    virtual int detectAudioConfiguration() const;
 
     static int paStreamCallback(const void *input,
                                 void *output,

app/streaming/audio/renderers/renderer.h

@@ -12,7 +12,6 @@ public:
 
     virtual bool prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig) = 0;
 
-    virtual void submitAudio(short* audioBuffer, int audioSize) = 0;
+    // Return false if an unrecoverable error has occurred and the renderer must be reinitialized
-
+    virtual bool submitAudio(short* audioBuffer, int audioSize) = 0;
-    virtual int detectAudioConfiguration() const = 0;
 };

app/streaming/audio/renderers/sdl.h

@@ -17,9 +17,7 @@ public:
 
     virtual bool prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig);
 
-    virtual void submitAudio(short* audioBuffer, int audioSize);
+    virtual bool submitAudio(short* audioBuffer, int audioSize);
-
-    virtual int detectAudioConfiguration() const;
 
 private:
     SDL_AudioDeviceID m_AudioDevice;

app/streaming/audio/renderers/sdlaud.cpp

@@ -10,47 +10,6 @@
 #define STOP_THE_WORLD_LIMIT 20
 #define DROP_RATIO_DENOM 32
 
-// This isn't accurate on macOS and Linux (PulseAudio),
-// since they both report supporting a large number of
-// channels, regardless of the actual output device.
-int SdlAudioRenderer::detectAudioConfiguration() const
-{
-    SDL_AudioSpec want, have;
-    SDL_AudioDeviceID dev;
-
-    SDL_zero(want);
-    want.freq = 48000;
-    want.format = AUDIO_S16;
-    want.channels = 6;
-    want.samples = 1024;
-
-    // Try to open for 5.1 surround sound, but allow SDL to tell us that's
-    // not available.
-    dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_CHANNELS_CHANGE);
-    if (dev == 0) {
-        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
-                     "Failed to open audio device");
-        // We'll probably have issues during audio stream init, but we'll
-        // try anyway
-        return AUDIO_CONFIGURATION_STEREO;
-    }
-
-    SDL_CloseAudioDevice(dev);
-
-    SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
-                "Audio device has %d channels", have.channels);
-
-    if (have.channels > 2) {
-        // We don't support quadraphonic or 7.1 surround, but SDL
-        // should be able to downmix or upmix better from 5.1 than
-        // from stereo, so use 5.1 in non-stereo cases.
-        return AUDIO_CONFIGURATION_51_SURROUND;
-    }
-    else {
-        return AUDIO_CONFIGURATION_STEREO;
-    }
-}
-
 SdlAudioRenderer::SdlAudioRenderer()
     : m_AudioDevice(0),
       m_ChannelCount(0),
@@ -141,7 +100,7 @@ SdlAudioRenderer::~SdlAudioRenderer()
     SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
 }
 
-void SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
+bool SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
 {
     m_SampleIndex++;
 
@@ -181,13 +140,13 @@ void SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
         // Hard drops happen all at once to forcefully
         // resync with the source.
         m_PendingHardDrops--;
-        return;
+        return true;
     }
     else if (m_PendingDrops != 0 && m_SampleIndex % DROP_RATIO_DENOM == 0) {
         // Normal drops are interspersed with the audio data
         // to hide the glitches.
         m_PendingDrops--;
-        return;
+        return true;
     }
 
     if (SDL_QueueAudio(m_AudioDevice, audioBuffer, audioSize) < 0) {
@@ -195,4 +154,6 @@ void SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
                      "Failed to queue audio sample: %s",
                      SDL_GetError());
     }
+
+    return true;
 }

app/streaming/session.cpp

@@ -288,7 +288,8 @@ Session::Session(NvComputer* computer, NvApp& app, StreamingPreferences *prefere
       m_DisplayOriginY(0),
       m_PendingWindowedTransition(false),
       m_OpusDecoder(nullptr),
-      m_AudioRenderer(nullptr)
+      m_AudioRenderer(nullptr),
+      m_AudioSampleCount(0)
 {
 }
 
@@ -327,10 +328,6 @@ void Session::initialize()
 
     switch (m_Preferences->audioConfig)
     {
-    case StreamingPreferences::AC_AUTO:
-        SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION, "Autodetecting audio configuration");
-        m_StreamConfig.audioConfiguration = detectAudioConfiguration();
-        break;
     case StreamingPreferences::AC_FORCE_STEREO:
         m_StreamConfig.audioConfiguration = AUDIO_CONFIGURATION_STEREO;
         break;

app/streaming/session.h

@@ -124,6 +124,7 @@ private:
     short m_OpusDecodeBuffer[MAX_CHANNELS * SAMPLES_PER_FRAME];
     IAudioRenderer* m_AudioRenderer;
     OPUS_MULTISTREAM_CONFIGURATION m_AudioConfig;
+    int m_AudioSampleCount;
 
     static AUDIO_RENDERER_CALLBACKS k_AudioCallbacks;
     static CONNECTION_LISTENER_CALLBACKS k_ConnCallbacks;