Support for AudioContextOptions latencyHint.

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>
 
 #include "base/command_line.h"
 #include "base/logging.h"
 #include "base/time/time.h"
 #include "build/build_config.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/render_frame_impl.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/base/silent_sink_suspender.h"
 #include "third_party/WebKit/public/web/WebLocalFrame.h"
 #include "third_party/WebKit/public/web/WebView.h"
 
 using blink::WebAudioDevice;
+using blink::WebAudioLatencyHint;
 using blink::WebLocalFrame;
 using blink::WebVector;
 using blink::WebView;
 
 namespace content {
 
+namespace {
+
+AudioDeviceFactory::SourceType GetLatencyHintSourceType(
+    WebAudioLatencyHint::Category latency_category) {
+  switch (latency_category) {
+    case WebAudioLatencyHint::CategoryInteractive:
+      return AudioDeviceFactory::kSourceWebAudioInteractive;
+    case WebAudioLatencyHint::CategoryBalanced:
+      return AudioDeviceFactory::kSourceWebAudioBalanced;
+    case WebAudioLatencyHint::CategoryPlayback:
+      return AudioDeviceFactory::kSourceWebAudioPlayback;
+  }
+  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
+  // starting the audio device.  The reason for all this is because the creator
+  // of the WebAudio objects might not be the actual source of the audio (e.g.,
+  // an extension creates a object that is passed and used within a page).
+  blink::WebLocalFrame* const web_frame =
+      blink::WebLocalFrame::frameForCurrentContext();
+  RenderFrame* const render_frame = RenderFrame::FromWebFrame(web_frame);
+  return render_frame ? render_frame->GetRoutingID() : MSG_ROUTING_NONE;
+}
+
+} // namespace
+
 RendererWebAudioDeviceImpl::RendererWebAudioDeviceImpl(
-    const media::AudioParameters& params,
+    media::ChannelLayout layout,
+    const blink::WebAudioLatencyHint& latency_hint,
     WebAudioDevice::RenderCallback* callback,
     int session_id,
     const url::Origin& security_origin)
-    : params_(params),
+    : latency_hint_(latency_hint),
       client_callback_(callback),
       session_id_(session_id),
       security_origin_(security_origin) {
   DCHECK(client_callback_);
+
+  media::AudioParameters hardware_params(
+      AudioDeviceFactory::GetOutputDeviceInfo(FrameIdFromCurrentContext(),
+                                              session_id_, std::string(),
+                                              security_origin_)
+          .output_params());
+
+  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();
+  }
+
+  DCHECK_NE(output_buffer_size, 0);
+
+  sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, layout,
+                     hardware_params.sample_rate(), 16, output_buffer_size);
+
+  // Specify the latency info to be passed to the browser side.
+  sink_params_.set_latency_tag(latency);
 }
 
 RendererWebAudioDeviceImpl::~RendererWebAudioDeviceImpl() {
   DCHECK(!sink_);
 }
 
 void RendererWebAudioDeviceImpl::start() {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   if (sink_)
     return;  // Already started.
 
-  // 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
-  // starting the audio device.  The reason for all this is because the creator
-  // of the WebAudio objects might not be the actual source of the audio (e.g.,
-  // an extension creates a object that is passed and used within a page).
-  WebLocalFrame* const web_frame = WebLocalFrame::frameForCurrentContext();
-  RenderFrame* const render_frame =
-      web_frame ? RenderFrame::FromWebFrame(web_frame) : NULL;
   sink_ = AudioDeviceFactory::NewAudioRendererSink(
-      AudioDeviceFactory::kSourceWebAudioInteractive,
-      render_frame ? render_frame->GetRoutingID() : MSG_ROUTING_NONE,
-      session_id_, std::string(), security_origin_);
-
-  // Specify the latency info to be passed to the browser side.
-  media::AudioParameters sink_params(params_);
-  sink_params.set_latency_tag(AudioDeviceFactory::GetSourceLatencyType(
-      AudioDeviceFactory::kSourceWebAudioInteractive));
+      GetLatencyHintSourceType(latency_hint_.category()),
+      FrameIdFromCurrentContext(), session_id_, std::string(),
+      security_origin_);
 
 #if defined(OS_ANDROID)
   // Use the media thread instead of the render thread for fake Render() calls
   // since it has special connotations for Blink and garbage collection. Timeout
   // value chosen to be highly unlikely in the normal case.
   webaudio_suspender_.reset(new media::SilentSinkSuspender(
-      this, base::TimeDelta::FromSeconds(30), sink_params, sink_,
+      this, base::TimeDelta::FromSeconds(30), sink_params_, sink_,
       RenderThreadImpl::current()->GetMediaThreadTaskRunner()));
-  sink_->Initialize(sink_params, webaudio_suspender_.get());
+  sink_->Initialize(sink_params_, webaudio_suspender_.get());
 #else
-  sink_->Initialize(sink_params, this);
+  sink_->Initialize(sink_params_, this);
 #endif
 
   sink_->Start();
   sink_->Play();
 }
 
 void RendererWebAudioDeviceImpl::stop() {
   DCHECK(thread_checker_.CalledOnValidThread());
   if (sink_) {
     sink_->Stop();
     sink_ = nullptr;
   }
 
 #if defined(OS_ANDROID)
   webaudio_suspender_.reset();
 #endif
 }
 
 double RendererWebAudioDeviceImpl::sampleRate() {
-  return params_.sample_rate();
+  return sink_params_.sample_rate();
+}
+
+int RendererWebAudioDeviceImpl::framesPerBuffer() {
+  return sink_params_.frames_per_buffer();
 }
 
 int RendererWebAudioDeviceImpl::Render(media::AudioBus* dest,
                                        uint32_t frames_delayed,
                                        uint32_t frames_skipped) {
   // Wrap the output pointers using WebVector.
   WebVector<float*> web_audio_dest_data(static_cast<size_t>(dest->channels()));
   for (int i = 0; i < dest->channels(); ++i)
     web_audio_dest_data[i] = dest->channel(i);
 
   // TODO(xians): Remove the following |web_audio_source_data| after
   // changing the blink interface.
   WebVector<float*> web_audio_source_data(static_cast<size_t>(0));
   client_callback_->render(web_audio_source_data, web_audio_dest_data,
                            dest->frames());
   return dest->frames();
 }
 
 void RendererWebAudioDeviceImpl::OnRenderError() {
   // TODO(crogers): implement error handling.
 }
 
 }  // namespace content