Support for AudioContextOptions latencyHint.

third_party/WebKit/Source/platform/audio/AudioDestination.cpp

Index: third_party/WebKit/Source/platform/audio/AudioDestination.cpp
diff --git a/third_party/WebKit/Source/platform/audio/AudioDestination.cpp b/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
index 39903c682010aa23536b1e7061ebce9d5a012e0b..803ad39bc11ccd7dfe4e8e2a01ae87382b3b69fe 100644
--- a/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
+++ b/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
@@ -34,6 +34,7 @@
 #include "platform/audio/AudioUtilities.h"
 #include "platform/weborigin/SecurityOrigin.h"
 #include "public/platform/Platform.h"
+#include "public/platform/WebAudioLatencyHint.h"
 #include "public/platform/WebSecurityOrigin.h"
 #include "wtf/PtrUtil.h"
 #include <memory>
@@ -49,18 +50,18 @@ std::unique_ptr<AudioDestination> AudioDestination::create(
     const String& inputDeviceId,
     unsigned numberOfInputChannels,
     unsigned numberOfOutputChannels,
-    float sampleRate,
+    const WebAudioLatencyHint& latencyHint,
     PassRefPtr<SecurityOrigin> securityOrigin) {
   return wrapUnique(new AudioDestination(
       callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels,
-      sampleRate, std::move(securityOrigin)));
+      latencyHint, std::move(securityOrigin)));
 }
 
 AudioDestination::AudioDestination(AudioIOCallback& callback,
                                    const String& inputDeviceId,
                                    unsigned numberOfInputChannels,
                                    unsigned numberOfOutputChannels,
-                                   float sampleRate,
+                                   const WebAudioLatencyHint& latencyHint,
                                    PassRefPtr<SecurityOrigin> securityOrigin)
     : m_callback(callback),
       m_numberOfOutputChannels(numberOfOutputChannels),
@@ -69,7 +70,6 @@ AudioDestination::AudioDestination(AudioIOCallback& callback,
       m_renderBus(AudioBus::create(numberOfOutputChannels,
                                    AudioUtilities::kRenderQuantumFrames,
                                    false)),
-      m_sampleRate(sampleRate),
       m_isPlaying(false) {
   // Histogram for audioHardwareBufferSize
   DEFINE_STATIC_LOCAL(SparseHistogram, hardwareBufferSizeHistogram,
@@ -80,41 +80,21 @@ AudioDestination::AudioDestination(AudioIOCallback& callback,
   DEFINE_STATIC_LOCAL(SparseHistogram, callbackBufferSizeHistogram,
                       ("WebAudio.AudioDestination.CallbackBufferSize"));
 
-  // Use the optimal buffer size recommended by the audio backend.
-  size_t recommendedHardwareBufferSize =
-      Platform::current()->audioHardwareBufferSize();
-  m_callbackBufferSize = recommendedHardwareBufferSize;
-
-#if OS(ANDROID)
-  // The optimum low-latency hardware buffer size is usually too small on
-  // Android for WebAudio to render without glitching. So, if it is small, use
-  // a larger size. If it was already large, use the requested size.
-  //
-  // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144
-  // for a Galaxy Nexus), cause significant processing jitter. Sometimes
-  // multiple blocks will processed, but other times will not be since the FIFO
-  // can satisfy the request. By using a larger callbackBufferSize, we smooth
-  // out the jitter.
-  const size_t kSmallBufferSize = 1024;
-  const size_t kDefaultCallbackBufferSize = 2048;
-
-  if (m_callbackBufferSize <= kSmallBufferSize)
-    m_callbackBufferSize = kDefaultCallbackBufferSize;
-#endif
+  m_audioDevice = wrapUnique(Platform::current()->createAudioDevice(
+      numberOfInputChannels, numberOfOutputChannels, latencyHint, this,
+      inputDeviceId, std::move(securityOrigin)));
+  DCHECK(m_audioDevice);
 
+  m_callbackBufferSize = m_audioDevice->framesPerBuffer();
   // Quick exit if the requested size is too large.
   DCHECK_LE(m_callbackBufferSize + AudioUtilities::kRenderQuantumFrames,
             fifoSize);
   if (m_callbackBufferSize + AudioUtilities::kRenderQuantumFrames > fifoSize)
     return;
 
-  m_audioDevice = wrapUnique(Platform::current()->createAudioDevice(
-      m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels,
-      sampleRate, this, inputDeviceId, std::move(securityOrigin)));
-  ASSERT(m_audioDevice);
-
   // Record the sizes if we successfully created an output device.
-  hardwareBufferSizeHistogram.sample(recommendedHardwareBufferSize);
+  hardwareBufferSizeHistogram.sample(
+      Platform::current()->audioHardwareBufferSize());
   callbackBufferSizeHistogram.sample(m_callbackBufferSize);

[o1ka, 2016/12/07 20:56:01]

rtoy@ - callback buffer size will be different for different latencies, won't
this histogram loose its meaning?
Is WebAudio interested in a separate histogram for each latency (keeping the
existing one for Interactive)?

[Raymond Toy, 2016/12/08 16:24:50]

Good point. It probably would be a good idea to add two more histograms to keep
track of the categories.  And an additional two more, maybe, to keep track of
the requested float latency and the actual value.

But we don't need to block this CL for that.  We have time to do that later. 

 
   // Create a FIFO to handle the possibility of the callback size