Add a silent audio sink to consume WebAudio data on silence detection.

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 "base/command_line.h"
 #include "base/logging.h"
+#include "base/single_thread_task_runner.h"
+#include "base/thread_task_runner_handle.h"
+#include "base/time/time.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/render_frame_impl.h"
 #include "media/audio/audio_output_device.h"
+#include "media/audio/null_audio_sink.h"
+#include "media/base/audio_timestamp_helper.h"
 #include "media/base/media_switches.h"
 #include "third_party/WebKit/public/web/WebLocalFrame.h"
 #include "third_party/WebKit/public/web/WebView.h"
 
 using blink::WebAudioDevice;
 using blink::WebLocalFrame;
 using blink::WebVector;
 using blink::WebView;
 
 namespace content {
 
+static const int kSilenceInSecondsToEnterIdleMode = 30.0;
+
 RendererWebAudioDeviceImpl::RendererWebAudioDeviceImpl(
     const media::AudioParameters& params,
     WebAudioDevice::RenderCallback* callback,
     int session_id)
     : params_(params),
       client_callback_(callback),
-      session_id_(session_id) {
+      session_id_(session_id),
+      audio_timestamp_helper_(
+          new media::AudioTimestampHelper(params.sample_rate())),
+      task_runner_(base::ThreadTaskRunnerHandle::Get()),
+      null_audio_sink_(new media::NullAudioSink(task_runner_)),
+      is_using_null_audio_sink_(false),
+      weak_factory_(this) {
   DCHECK(client_callback_);
+  audio_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
+  null_audio_sink_->Initialize(params_, this);
+  weak_this_ = weak_factory_.GetWeakPtr();
 }
 
 RendererWebAudioDeviceImpl::~RendererWebAudioDeviceImpl() {
-  DCHECK(!output_device_.get());
+  DCHECK(!output_device_);
 }
 
 void RendererWebAudioDeviceImpl::start() {
   DCHECK(thread_checker_.CalledOnValidThread());
 
-  if (output_device_.get())
+  if (output_device_)
     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;
   output_device_ = AudioDeviceFactory::NewOutputDevice(
       render_frame ? render_frame->GetRoutingID(): MSG_ROUTING_NONE);
   output_device_->InitializeWithSessionId(params_, this, session_id_);
   output_device_->Start();
   // Note: Default behavior is to auto-play on start.
 }
 
 void RendererWebAudioDeviceImpl::stop() {
   DCHECK(thread_checker_.CalledOnValidThread());
 
-  if (output_device_.get()) {
+  if (output_device_) {
     output_device_->Stop();
     output_device_ = NULL;
   }
+  StopNullAudioSink();

[DaleCurtis, 2015/06/22 19:14:40]

Stop can be called multiple times, so I'd just call it directly.

[qinmin, 2015/06/22 22:23:16]

Done.

 }
 
 double RendererWebAudioDeviceImpl::sampleRate() {
   return params_.sample_rate();
 }
 
 int RendererWebAudioDeviceImpl::Render(media::AudioBus* dest,
                                        int audio_delay_milliseconds) {
   if (client_callback_) {
     // 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());
   }
 
+#if defined(OS_ANDROID)
+  if (!dest->AreFramesZero() && is_using_null_audio_sink_) {

[DaleCurtis, 2015/06/22 19:14:40]

We lose this buffer in this case, do we want to save it for the next Render()
call or just drop it?

[qinmin, 2015/06/22 22:23:16]

I had the same question here. Changed the code to have an extra  AudioBus to
store the data.

+    // This is called on the main render thread when audio is detected.
+    output_device_->Play();
+    is_using_null_audio_sink_ = false;
+    task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(&RendererWebAudioDeviceImpl::StopNullAudioSink,
+                   weak_this_));
+  } else if (!is_using_null_audio_sink_) {
+    // Called on the audio device thread.
+    audio_timestamp_helper_->AddFrames(dest->frames());

[DaleCurtis, 2015/06/22 19:14:40]

Instead of using a timestamp helper:

On, l.108:
const bool is_zero = dest->AreFramesZero();
if (!is_zero)
  first_silence_time_ = base::TimeTicks();

Within l.117:
const base::TimeTicks now = base::TimeTicks::Now();
if (first_silence_time_.is_null())
  first_silence_time_ = now;

if (now - first_silence_time_ > kAllowedSilence) {
  ...
}

To use this in unittests you'll want to use a base::TickClock and
base::SimpleTestTickClock, see WallClockTimeSourceTest for an example.

[qinmin, 2015/06/22 22:23:16]

If I remember clearly, TimtTicks::Now() are very expensive on Android. A
timestamp helper just use the math of addition and multiplication, so it is much
cheaper.
And I use AudioTimestampHelper on l.130 to calculate when NullAudioSink should
be started.
Anyway, modified the code according to your suggestion, but it looks to me the
code looks much heavier.

[DaleCurtis, 2015/06/22 22:49:42]

Do you have a source for TimeTicks::Now being expensive on android? I don't
think that's true anymore. It's just a call to clock_gettime() which a brief
search doesn't reveal slowness on android for.

[qinmin, 2015/06/23 00:01:53]

see
http://gamasutra.com/view/feature/171774/getting_high_precision_timing_on_.ph....
This involves a syscall and it can be expensive on many platforms

[DaleCurtis, 2015/06/23 16:38:14]

Despite those early statements, the benchmarks show the Java method nanoTime(),
based on clock_gettime(), completing 100k runs in 892ns, I can only imagine the
native call is even faster, so I think we're okay :)

+    if (audio_timestamp_helper_->GetTimestamp().InSecondsF()
+        > kSilenceInSecondsToEnterIdleMode) {
+      output_device_->Pause();
+      audio_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
+      is_using_null_audio_sink_ = true;
+      // If Stop() is called right after the task is posted,
+      // StartNullAudioSink() should do nothing.
+      task_runner_->PostDelayedTask(
+          FROM_HERE,
+          base::Bind(&RendererWebAudioDeviceImpl::StartNullAudioSink,
+                     weak_this_),
+          audio_timestamp_helper_->GetFrameDuration(dest->frames()));
+    }
+  }
+#endif
   return dest->frames();
 }
 
 void RendererWebAudioDeviceImpl::OnRenderError() {
   // TODO(crogers): implement error handling.
 }
 
+void RendererWebAudioDeviceImpl::StopNullAudioSink() {
+  if (null_audio_sink_)

[DaleCurtis, 2015/06/22 19:14:41]

Shouldn't this never be null? Additionally since AudioRendererSink objects are
ref-counted you can just bind directly to the null sink instead of using a
weak_this_ and helper methods.

[qinmin, 2015/06/22 22:23:16]

Done.

+    null_audio_sink_->Stop();
+}
+
+void RendererWebAudioDeviceImpl::StartNullAudioSink() {
+  if (null_audio_sink_)
+    null_audio_sink_->Play();
+}
+
 }  // namespace content