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 4cccdd9a4682f2157244b6b2ac7c8a6278c20819..8882377a67983c4c1fc322b4cd365ebf14d801dd 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),

[Raymond Toy, 2016/11/15 18:16:17]

Where is this initialized now?  It's really important.

[Andrew MacPherson, 2016/11/16 10:58:30]

Based on an earlier suggestion from o1ka@ I changed it so that the sample rate
is not specified on the Blink side but can instead be queried from the
WebAudioDevice::sampleRate(). It's possible that I have misunderstood though so
let me know if so!

       m_isPlaying(false) {
   // Histogram for audioHardwareBufferSize
   DEFINE_STATIC_LOCAL(SparseHistogram, hardwareBufferSizeHistogram,
@@ -80,41 +80,20 @@ 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);

[hongchan, 2016/11/15 23:46:01]

So I guess now we have to query the sample rate via m_audioDevice. Is this your
intention?

[Andrew MacPherson, 2016/11/16 10:58:29]

Yes, this was the intention but I will let o1ka@ confirm.

 
+  m_callbackBufferSize = m_audioDevice->framesPerBuffer();

[hongchan, 2016/11/15 23:46:01]

o1ka@

If we do this, is this going to be an IPC call? Similarly, will
|m_audioDevice->sampleRate()| be an IPC call as well?

[o1ka, 2016/11/17 17:37:25]

No extra IPC, the value is cached in audio device. Overall, the change does not
introduce any extra IPC, it just delays a call to fetch device parameters until
audio device is constructed (and thus we can ask for specific device parameters
instead of the default ones)

   // 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(m_callbackBufferSize);

[o1ka, 2016/11/15 22:44:28]

I'm not sure if these two histograms have the right meaning now

[hongchan, 2016/11/15 23:46:01]

I think o1ka is right. This seems wrong and we might have to remove the first
one since we don't have that info in the Blink boundary any more.

[Raymond Toy, 2016/11/15 23:50:38]

Yeah, but I think we really want this somewhere. It's really useful information,
especially on Android, where, I think, we are extremely conservative in the
actual buffer size.  This needs to be optimized more (in a different CL).

[Andrew MacPherson, 2016/11/16 10:58:30]

Let me know if you'd like me to move this elsewhere (to the 'content' side?) or
just remove it entirely and I can take care of that.

   callbackBufferSizeHistogram.sample(m_callbackBufferSize);
 
   // Create a FIFO to handle the possibility of the callback size