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 44 matching lines @@
       client_callback_(callback),
       session_id_(session_id),
       security_origin_(security_origin),
       frame_id_(render_frame_id_cb.Run()) {
   DCHECK(client_callback_);
   DCHECK_NE(frame_id_, MSG_ROUTING_NONE);
 
   media::AudioParameters hardware_params(device_params_cb.Run(
       frame_id_, session_id_, std::string(), security_origin_));
 
   int output_buffer_size = 0;

[o1ka, 2017/03/16 10:51:23]

Would be nice to wrap |output_buffer_size| calculation into a helper function
GetOutputBufferSize(latency, hardware_params)

[Andrew MacPherson, 2017/03/16 17:02:09]

Done, I had to pass the latency_hint as well since we need to access its
.seconds() method if its type is 'exact', even though this is a bit redundant.
The alternative is to create 'latency' in both the constructor and
GetOutputBufferSize since it's needed in both places.

 
   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.
+      output_buffer_size = output_buffer_size =

[o1ka, 2017/03/16 10:51:23]

extra "output_buffer_size ="?

[Andrew MacPherson, 2017/03/16 17:02:09]

Oops, fixed!

+          media::AudioLatency::GetExactBufferSize(
+              latency_hint.seconds() * hardware_params.sample_rate(),
+              media::AudioLatency::GetHighLatencyBufferSize(
+                  hardware_params.sample_rate(), 0),
+              hardware_params.frames_per_buffer());

[o1ka, 2017/03/16 10:51:24]

Can it be GetExactBufferSize(duration, hardware_params)?

[Andrew MacPherson, 2017/03/16 17:02:09]

Done.

+      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);

[o1ka, 2017/03/16 10:51:24]

We need top-to-bottom browser tests to see if a platform can deal with certain
buffer sizes and how audio quality looks like.

[Andrew MacPherson, 2017/03/16 17:02:09]

Do you mean like a renderer_webaudiodevice_impl_browsertest.cc or something
else? I can certainly look into adding this and making it try unusual buffer
sizes, I might need some pointers around adding checks for audio quality though.

[o1ka, 2017/03/17 11:17:22]

Let's see what dalecurtis@ thinks about testing it.

   // 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.
   sink_params_.set_channels_for_discrete(channels);
 
   // Specify the latency info to be passed to the browser side.
   sink_params_.set_latency_tag(latency);
 }
 
 RendererWebAudioDeviceImpl::~RendererWebAudioDeviceImpl() {
   DCHECK(!sink_);
@@ skipping 80 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