Support AudioContextOptions latencyHint as double.

content/renderer/media/renderer_webaudiodevice_impl.cc

Index: content/renderer/media/renderer_webaudiodevice_impl.cc
diff --git a/content/renderer/media/renderer_webaudiodevice_impl.cc b/content/renderer/media/renderer_webaudiodevice_impl.cc
index 2c4f81b185d74fe96ab0a25d677a67fd0d3d9296..8eb0b240b9004af852bd3ff0874a69f4118f8d79 100644
--- a/content/renderer/media/renderer_webaudiodevice_impl.cc
+++ b/content/renderer/media/renderer_webaudiodevice_impl.cc
@@ -41,13 +41,39 @@ AudioDeviceFactory::SourceType GetLatencyHintSourceType(
     case WebAudioLatencyHint::kCategoryPlayback:
       return AudioDeviceFactory::kSourceWebAudioPlayback;
     case WebAudioLatencyHint::kCategoryExact:
-      // TODO implement CategoryExact
-      return AudioDeviceFactory::kSourceWebAudioInteractive;
+      return AudioDeviceFactory::kSourceWebAudioExact;
   }
   NOTREACHED();
   return AudioDeviceFactory::kSourceWebAudioInteractive;
 }
 
+int GetOutputBufferSize(const blink::WebAudioLatencyHint& latency_hint,
+                        const media::AudioLatency::LatencyType latency,
+                        const media::AudioParameters& hardware_params) {
+  // Adjust output buffer size according to the latency requirement.
+  switch (latency) {
+    case media::AudioLatency::LATENCY_INTERACTIVE:
+      return media::AudioLatency::GetInteractiveBufferSize(
+          hardware_params.frames_per_buffer());
+      break;
+    case media::AudioLatency::LATENCY_RTC:
+      return media::AudioLatency::GetRtcBufferSize(
+          hardware_params.sample_rate(), hardware_params.frames_per_buffer());
+      break;
+    case media::AudioLatency::LATENCY_PLAYBACK:
+      return media::AudioLatency::GetHighLatencyBufferSize(

[DaleCurtis, 2017/03/28 17:34:07]

I wonder if we should just replace this with GetExactBufferSize(0.2s, params).
We can do that in a followup though.

[Andrew MacPherson, 2017/03/29 08:34:09]

Sounds good!

+          hardware_params.sample_rate(), 0);
+      break;
+    case media::AudioLatency::LATENCY_EXACT_MS:
+      return media::AudioLatency::GetExactBufferSize(latency_hint.seconds(),
+                                                     hardware_params);
+      break;
+    default:
+      NOTREACHED();
+  }
+  return 0;
+}
+
 int FrameIdFromCurrentContext() {
   // Assumption: This method is being invoked within a V8 call stack.  CHECKs
   // will fail in the call to frameForCurrentContext() otherwise.
@@ -107,32 +133,11 @@ RendererWebAudioDeviceImpl::RendererWebAudioDeviceImpl(
   media::AudioParameters hardware_params(device_params_cb.Run(
       frame_id_, session_id_, std::string(), security_origin_));
 
-  int output_buffer_size = 0;
-
   media::AudioLatency::LatencyType latency =
       AudioDeviceFactory::GetSourceLatencyType(
-          GetLatencyHintSourceType(latency_hint_.category()));
-
-  // Adjust output buffer size according to the latency requirement.
-  switch (latency) {
-    case media::AudioLatency::LATENCY_INTERACTIVE:
-      output_buffer_size = media::AudioLatency::GetInteractiveBufferSize(
-          hardware_params.frames_per_buffer());
-      break;
-    case media::AudioLatency::LATENCY_RTC:
-      output_buffer_size = media::AudioLatency::GetRtcBufferSize(
-          hardware_params.sample_rate(), hardware_params.frames_per_buffer());
-      break;
-    case media::AudioLatency::LATENCY_PLAYBACK:
-      output_buffer_size = media::AudioLatency::GetHighLatencyBufferSize(
-          hardware_params.sample_rate(), 0);
-      break;
-    case media::AudioLatency::LATENCY_EXACT_MS:
-    // TODO(olka): add support when WebAudio requires it.
-    default:
-      NOTREACHED();
-  }
-
+          GetLatencyHintSourceType(latency_hint.category()));
+  int output_buffer_size =
+      GetOutputBufferSize(latency_hint_, latency, hardware_params);
   DCHECK_NE(output_buffer_size, 0);
 
   sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, layout,