Support for AudioContextOptions latencyHint.

content/renderer/renderer_blink_platform_impl.cc

Index: content/renderer/renderer_blink_platform_impl.cc
diff --git a/content/renderer/renderer_blink_platform_impl.cc b/content/renderer/renderer_blink_platform_impl.cc
index 6bdaa6c4e881e57086e69eaf641f650e636447f5..f190bae7233a23187a331c1050be7660256f5636 100644
--- a/content/renderer/renderer_blink_platform_impl.cc
+++ b/content/renderer/renderer_blink_platform_impl.cc
@@ -87,6 +87,7 @@
 #include "third_party/WebKit/public/platform/BlameContext.h"
 #include "third_party/WebKit/public/platform/FilePathConversion.h"
 #include "third_party/WebKit/public/platform/URLConversion.h"
+#include "third_party/WebKit/public/platform/WebAudioLatencyHint.h"
 #include "third_party/WebKit/public/platform/WebBlobRegistry.h"
 #include "third_party/WebKit/public/platform/WebDeviceLightListener.h"
 #include "third_party/WebKit/public/platform/WebFileInfo.h"
@@ -142,6 +143,7 @@
 
 using blink::Platform;
 using blink::WebAudioDevice;
+using blink::WebAudioLatencyHint;
 using blink::WebBlobRegistry;
 using blink::WebCanvasCaptureHandler;
 using blink::WebDatabaseObserver;
@@ -657,16 +659,17 @@ WebDatabaseObserver* RendererBlinkPlatformImpl::databaseObserver() {
 }
 
 WebAudioDevice* RendererBlinkPlatformImpl::createAudioDevice(
-    size_t buffer_size,
     unsigned input_channels,
     unsigned channels,
-    double sample_rate,
+    const blink::WebAudioLatencyHint& latency_hint,
     WebAudioDevice::RenderCallback* callback,
     const blink::WebString& input_device_id,
     const blink::WebSecurityOrigin& security_origin) {
   // Use a mock for testing.
   blink::WebAudioDevice* mock_device =
-      GetContentClient()->renderer()->OverrideCreateAudioDevice(sample_rate);
+      GetContentClient()->renderer()->OverrideCreateAudioDevice(
+          GetAudioHardwareParams().sample_rate(),

[o1ka, 2016/11/30 11:46:26]

Make OverrideCreateAudioDevice() parameterless?
It's up to implementation to decide whether is needs real or mock sample
rate/buffer size.

[Andrew MacPherson, 2016/12/01 12:11:56]

rtoy@: Do you know if we can just hardcode these to something reasonable in
layout_test_content_renderer_client.cc or if we need to retrieve actual hardware
values?

[Raymond Toy, 2016/12/01 16:16:27]

I don't think any of the build bots have any actual audio hardware (real or
virtual) so there's nothing to be retrieved.  This is why none of the layout
tests actually use AudioContext because it doesn't really work.

[Andrew MacPherson, 2016/12/02 09:42:55]

Ok, I've removed the parameters from OverrideCreateAudioDevice() and have
hardcoded them to 44100 and 128 in layout_test_content_renderer_client.cc.

+          GetAudioHardwareParams().frames_per_buffer());
   if (mock_device)
     return mock_device;
 
@@ -700,9 +703,9 @@ WebAudioDevice* RendererBlinkPlatformImpl::createAudioDevice(
       layout = media::CHANNEL_LAYOUT_7_1;
       break;
     default:
-      // If the layout is not supported (more than 9 channels), falls back to
-      // discrete mode.
-      layout = media::CHANNEL_LAYOUT_DISCRETE;
+      // TODO need to also pass 'channels' into RendererWebAudioDeviceImpl for
+      // CHANNEL_LAYOUT_DISCRETE
+      NOTREACHED();
   }
 
   int session_id = 0;
@@ -715,15 +718,9 @@ WebAudioDevice* RendererBlinkPlatformImpl::createAudioDevice(
     input_channels = 0;
   }
 
-  // For CHANNEL_LAYOUT_DISCRETE, pass the explicit channel count along with
-  // the channel layout when creating an |AudioParameters| object.
-  media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                                layout, static_cast<int>(sample_rate), 16,
-                                buffer_size);
-  params.set_channels_for_discrete(channels);
-
   return new RendererWebAudioDeviceImpl(
-      params, callback, session_id, static_cast<url::Origin>(security_origin));
+      layout, latency_hint, callback, session_id,
+      static_cast<url::Origin>(security_origin));
 }
 
 bool RendererBlinkPlatformImpl::loadAudioResource(