Use queued audio duration instead of queued frame count to constrain latency

This avoids latency explosion when we negotiate 10 ms instead of 5 ms audio packets.

Author: cgutman

app/streaming/audio/renderers/sdl.h

@@ -21,5 +21,6 @@ class SdlAudioRenderer : public IAudioRenderer
 private:
     SDL_AudioDeviceID m_AudioDevice;
     void* m_AudioBuffer;
-    int m_FrameSize;
+    Uint32 m_FrameSize;
+    Uint32 m_FrameDurationMs;
 };

app/streaming/audio/renderers/sdlaud.cpp

@@ -32,6 +32,7 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
     // The buffering helps avoid audio underruns due to network jitter.
     want.samples = SDL_max(480, opusConfig->samplesPerFrame * 3);
 
+    m_FrameDurationMs = opusConfig->samplesPerFrame / (opusConfig->sampleRate / 1000);
     m_FrameSize = opusConfig->samplesPerFrame *
                   opusConfig->channelCount *
                   getAudioBufferSampleSize();
@@ -115,8 +116,8 @@ bool SdlAudioRenderer::submitAudio(int bytesWritten)
             return false;
         }
 
-        // Only queue more samples where there are 10 frames or less in SDL's queue
-        if (SDL_GetQueuedAudioSize(m_AudioDevice) / m_FrameSize <= 10) {
+        // Only queue more samples where there is 50 ms or less in SDL's queue
+        if (SDL_GetQueuedAudioSize(m_AudioDevice) / m_FrameSize * m_FrameDurationMs <= 50) {
             break;
         }