Support AudioContextOptions latencyHint as double.

content/renderer/media/renderer_webaudiodevice_impl.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/renderer_webaudiodevice_impl.h"
 
 #include <stddef.h>
 
 #include <string>
 
@@ skipping 23 matching lines @@
 AudioDeviceFactory::SourceType GetLatencyHintSourceType(
     WebAudioLatencyHint::AudioContextLatencyCategory latency_category) {
   switch (latency_category) {
     case WebAudioLatencyHint::kCategoryInteractive:
       return AudioDeviceFactory::kSourceWebAudioInteractive;
     case WebAudioLatencyHint::kCategoryBalanced:
       return AudioDeviceFactory::kSourceWebAudioBalanced;
     case WebAudioLatencyHint::kCategoryPlayback:
       return AudioDeviceFactory::kSourceWebAudioPlayback;
     case WebAudioLatencyHint::kCategoryExact:
-      // TODO implement CategoryExact
-      return AudioDeviceFactory::kSourceWebAudioInteractive;
+      return AudioDeviceFactory::kSourceWebAudioExact;
   }
   NOTREACHED();
   return AudioDeviceFactory::kSourceWebAudioInteractive;
 }
 
 int FrameIdFromCurrentContext() {
   // Assumption: This method is being invoked within a V8 call stack.  CHECKs
   // will fail in the call to frameForCurrentContext() otherwise.
   //
   // Therefore, we can perform look-ups to determine which RenderView is
@@ skipping 65 matching lines @@
       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.
+      output_buffer_size =
+          latency_hint.seconds() * hardware_params.sample_rate(),
+      output_buffer_size =
+          std::max(std::min(output_buffer_size,
+                            media::AudioLatency::GetHighLatencyBufferSize(
+                                hardware_params.sample_rate(), 0)),
+                   hardware_params.frames_per_buffer());

[o1ka, 2017/03/14 15:55:43]

Looks like we need something like media::AudioLatency::GetExactBufferSize()?

[Andrew MacPherson, 2017/03/15 15:08:17]

Done, I've added this now taking a parameter for the preferred size, min, and
max, let me know if that makes sense!

+      break;
     default:
       NOTREACHED();
   }
 
   DCHECK_NE(output_buffer_size, 0);
 
   sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, layout,
                      hardware_params.sample_rate(), 16, output_buffer_size);
   // Always set channels, this should be a no-op in all but the discrete case;
   // this call will fail if channels doesn't match the layout in other cases.
@@ skipping 88 matching lines @@
 const scoped_refptr<base::SingleThreadTaskRunner>&
 RendererWebAudioDeviceImpl::GetMediaTaskRunner() {
   if (!media_task_runner_) {
     media_task_runner_ =
         RenderThreadImpl::current()->GetMediaThreadTaskRunner();
   }
   return media_task_runner_;
 }
 
 }  // namespace content