Bug 907817 - Allow AudioStream users to pass-in latency requirements. r=kinetik

--HG--
extra : rebase_source : 3cf1971913b3cfd8000c58614b0d0be8b7805af0

content/html/content/src/HTMLAudioElement.cpp

@@ -114,7 +114,7 @@ HTMLAudioElement::MozSetup(uint32_t aChannels, uint32_t aRate, ErrorResult& aRv)
 #endif
 
   mAudioStream = AudioStream::AllocateStream();
-  aRv = mAudioStream->Init(aChannels, aRate, mAudioChannelType);
+  aRv = mAudioStream->Init(aChannels, aRate, mAudioChannelType, AudioStream::HighLatency);
   if (aRv.Failed()) {
     mAudioStream->Shutdown();
     mAudioStream = nullptr;

content/media/AudioStream.cpp

@@ -41,6 +41,8 @@ PRLogModuleInfo* gAudioStreamLog = nullptr;
 static Mutex* gAudioPrefsLock = nullptr;
 static double gVolumeScale;
 static uint32_t gCubebLatency;
+static bool gCubebLatencyPrefSet;
+static const uint32_t CUBEB_NORMAL_LATENCY_MS = 100;
 
 /**
  * When MOZ_DUMP_AUDIO is set in the environment (to anything),
@@ -65,9 +67,10 @@ static int PrefChanged(const char* aPref, void* aClosure)
     // Arbitrary default stream latency of 100ms.  The higher this
     // value, the longer stream volume changes will take to become
     // audible.
-    uint32_t value = Preferences::GetUint(aPref, 100);
+    gCubebLatencyPrefSet = Preferences::HasUserValue(aPref);
+    uint32_t value = Preferences::GetUint(aPref, CUBEB_NORMAL_LATENCY_MS);
     MutexAutoLock lock(*gAudioPrefsLock);
-    gCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 20), 1000);
+    gCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 1), 1000);
   }
   return 0;
 }
@@ -97,6 +100,12 @@ static uint32_t GetCubebLatency()
   MutexAutoLock lock(*gAudioPrefsLock);
   return gCubebLatency;
 }
+
+static bool CubebLatencyPrefSet()
+{
+  MutexAutoLock lock(*gAudioPrefsLock);
+  return gCubebLatencyPrefSet;
+}
 #endif
 
 #if defined(MOZ_CUBEB) && defined(__ANDROID__) && defined(MOZ_B2G)
@@ -311,7 +320,8 @@ class BufferedAudioStream : public AudioStream
   ~BufferedAudioStream();
 
   nsresult Init(int32_t aNumChannels, int32_t aRate,
-                const dom::AudioChannelType aAudioChannelType);
+                const dom::AudioChannelType aAudioChannelType,
+                AudioStream::LatencyRequest aLatencyRequest);
   void Shutdown();
   nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames);
   uint32_t Available();
@@ -526,7 +536,8 @@ BufferedAudioStream::EnsureTimeStretcherInitialized()
 
 nsresult
 BufferedAudioStream::Init(int32_t aNumChannels, int32_t aRate,
-                            const dom::AudioChannelType aAudioChannelType)
+                          const dom::AudioChannelType aAudioChannelType,
+                          AudioStream::LatencyRequest aLatencyRequest)
 {
   cubeb* cubebContext = GetCubebContext();
 
@@ -562,10 +573,22 @@ BufferedAudioStream::Init(int32_t aNumChannels, int32_t aRate,
 
   mAudioClock.Init();
 
+  // If the latency pref is set, use it. Otherwise, if this stream is intended
+  // for low latency playback, try to get the lowest latency possible.
+  // Otherwise, for normal streams, use 100ms.
+  uint32_t latency;
+  if (aLatencyRequest == AudioStream::LowLatency && !CubebLatencyPrefSet()) {
+    if (cubeb_get_min_latency(cubebContext, params, &latency) != CUBEB_OK) {
+      latency = GetCubebLatency();
+    }
+  } else {
+    latency = GetCubebLatency();
+  }
+
   {
     cubeb_stream* stream;
     if (cubeb_stream_init(cubebContext, &stream, "BufferedAudioStream", params,
-                          GetCubebLatency(), DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
+                          latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
       mCubebStream.own(stream);
     }
   }

content/media/AudioStream.h

@@ -92,6 +92,10 @@ class AudioClock
 class AudioStream
 {
 public:
+  enum LatencyRequest {
+    HighLatency,
+    LowLatency
+  };
   AudioStream();
 
   virtual ~AudioStream();
@@ -116,7 +120,8 @@ public:
   // channels (1 for mono, 2 for stereo, etc) and aRate is the sample rate
   // (22050Hz, 44100Hz, etc).
   virtual nsresult Init(int32_t aNumChannels, int32_t aRate,
-                        const dom::AudioChannelType aAudioStreamType) = 0;
+                        const dom::AudioChannelType aAudioStreamType,
+                        LatencyRequest aLatencyRequest) = 0;
 
   // Closes the stream. All future use of the stream is an error.
   virtual void Shutdown() = 0;

content/media/MediaDecoderStateMachine.cpp

@@ -1059,7 +1059,7 @@ void MediaDecoderStateMachine::AudioLoop()
     // circumstances, so we take care to drop the decoder monitor while
     // initializing.
     nsAutoPtr<AudioStream> audioStream(AudioStream::AllocateStream());
-    audioStream->Init(channels, rate, audioChannelType);
+    audioStream->Init(channels, rate, audioChannelType, AudioStream::HighLatency);
     audioStream->SetVolume(volume);
     if (audioStream->SetPreservesPitch(preservesPitch) != NS_OK) {
       NS_WARNING("Setting the pitch preservation failed at AudioLoop start.");

content/media/MediaStreamGraph.cpp

@@ -768,7 +768,7 @@ MediaStreamGraphImpl::CreateOrDestroyAudioStreams(GraphTime aAudioOutputStartTim
         audioOutputStream->mStream = AudioStream::AllocateStream();
         // XXX for now, allocate stereo output. But we need to fix this to
         // match the system's ideal channel configuration.
-        audioOutputStream->mStream->Init(2, tracks->GetRate(), AUDIO_CHANNEL_NORMAL);
+        audioOutputStream->mStream->Init(2, tracks->GetRate(), AUDIO_CHANNEL_NORMAL, AudioStream::LowLatency);
         audioOutputStream->mTrackID = tracks->GetID();
       }
     }