Improve low bandwidth audio performance for SDL and SLAudio and fix RTSP issues with broken PMTUD

app/streaming/audio/renderers/sdl.h

@@ -21,6 +21,5 @@ public:
 private:
     SDL_AudioDeviceID m_AudioDevice;
     void* m_AudioBuffer;
-    int m_FrameDurationMs;
     int m_FrameSize;
 };

app/streaming/audio/renderers/sdlaud.cpp

@@ -34,7 +34,6 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
     want.samples = opusConfig->samplesPerFrame;
 
     m_FrameSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;
-    m_FrameDurationMs = opusConfig->samplesPerFrame / 48;
 
     m_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
     if (m_AudioDevice == 0) {
@@ -97,7 +96,7 @@ bool SdlAudioRenderer::submitAudio(int bytesWritten)
 
     // Don't queue if there's already more than 30 ms of audio data waiting
     // in Moonlight's audio queue.
-    if (LiGetPendingAudioFrames() * m_FrameDurationMs > 30) {
+    if (LiGetPendingAudioDuration() > 30) {
         return true;
     }
 
@@ -118,6 +117,6 @@ bool SdlAudioRenderer::submitAudio(int bytesWritten)
 
 int SdlAudioRenderer::getCapabilities()
 {
-    // Direct submit can't be used because we use LiGetPendingAudioFrames()
+    // Direct submit can't be used because we use LiGetPendingAudioDuration()
-    return 0;
+    return CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
 }

app/streaming/audio/renderers/slaud.cpp

@@ -23,7 +23,6 @@ bool SLAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* o
     // it's hard to avoid since we get crushed by CPU limitations.
     m_MaxQueuedAudioMs = 40 * opusConfig->channelCount / 2;
 
-    m_FrameDuration = opusConfig->samplesPerFrame / 48;
     m_AudioBufferSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;
     m_AudioStream = SLAudio_CreateStream(m_AudioContext,
                                          opusConfig->sampleRate,
@@ -37,8 +36,8 @@ bool SLAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* o
     }
 
     SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
-                "Using SLAudio renderer with %d ms frames",
+                "Using SLAudio renderer with %d samples per frame",
-                m_FrameDuration);
+                opusConfig->samplesPerFrame);
 
     return true;
 }
@@ -100,7 +99,7 @@ bool SLAudioRenderer::submitAudio(int bytesWritten)
         return true;
     }
 
-    if (LiGetPendingAudioFrames() * m_FrameDuration < m_MaxQueuedAudioMs) {
+    if (LiGetPendingAudioDuration() < m_MaxQueuedAudioMs) {
         SLAudio_SubmitFrame(m_AudioStream);
         m_AudioBuffer = nullptr;
     }
@@ -115,7 +114,7 @@ bool SLAudioRenderer::submitAudio(int bytesWritten)
 
 int SLAudioRenderer::getCapabilities()
 {
-    return CAPABILITY_SLOW_OPUS_DECODER;
+    return CAPABILITY_SLOW_OPUS_DECODER | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
 }
 
 void SLAudioRenderer::slLogCallback(void*, ESLAudioLog logLevel, const char *message)

app/streaming/audio/renderers/slaud.h

@@ -28,6 +28,5 @@ private:
 
     void* m_AudioBuffer;
     int m_AudioBufferSize;
-    int m_FrameDuration;
     int m_MaxQueuedAudioMs;
 };