Switching audio clients to using RestartableAudioRendererSink interface as a sink.

content/renderer/media/track_audio_renderer.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/track_audio_renderer.h"
 
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/synchronization/lock.h"
 #include "base/thread_task_runner_handle.h"
 #include "base/trace_event/trace_event.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_track.h"
-#include "media/audio/audio_output_device.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_shifter.h"
 
 namespace content {
 
 namespace {
 
 enum LocalRendererSinkStates {
   kSinkStarted = 0,
   kSinkNeverStarted,
@@ skipping 118 matching lines @@
 
 void TrackAudioRenderer::Start() {
   DVLOG(1) << "TrackAudioRenderer::Start()";
   DCHECK(task_runner_->BelongsToCurrentThread());
   DCHECK_EQ(playing_, false);
 
   // We get audio data from |audio_track_|...
   MediaStreamAudioSink::AddToAudioTrack(this, audio_track_);
   // ...and |sink_| will get audio data from us.
   DCHECK(!sink_.get());
-  sink_ =
-      AudioDeviceFactory::NewOutputDevice(playout_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
+  sink_ = AudioDeviceFactory::NewAudioRendererSink(
+      AudioDeviceFactory::kSourceNonRtcAudioTrack, playout_render_frame_id_,
+      session_id_, output_device_id_, security_origin_);
 
   base::AutoLock auto_lock(thread_lock_);
   prior_elapsed_render_time_ = base::TimeDelta();
   num_samples_rendered_ = 0;
 }
 
 void TrackAudioRenderer::Stop() {
   DVLOG(1) << "TrackAudioRenderer::Stop()";
   DCHECK(task_runner_->BelongsToCurrentThread());
 
@@ skipping 79 matching lines @@
     const url::Origin& security_origin,
     const media::SwitchOutputDeviceCB& callback) {
   DVLOG(1) << "TrackAudioRenderer::SwitchOutputDevice()";
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   {
     base::AutoLock auto_lock(thread_lock_);
     HaltAudioFlowWhileLockHeld();
   }
 
-  scoped_refptr<media::AudioOutputDevice> new_sink =
-      AudioDeviceFactory::NewOutputDevice(playout_render_frame_id_, session_id_,
-                                          device_id, security_origin);
-  if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) {
-    callback.Run(new_sink->GetDeviceStatus());
+  scoped_refptr<media::AudioRendererSink> new_sink =
+      AudioDeviceFactory::NewAudioRendererSink(
+          AudioDeviceFactory::kSourceNonRtcAudioTrack, playout_render_frame_id_,
+          session_id_, device_id, security_origin);
+  media::OutputDeviceStatus new_sink_status =
+      new_sink->GetOutputDevice()->GetDeviceStatus();
+  if (new_sink_status != media::OUTPUT_DEVICE_STATUS_OK) {
+    callback.Run(new_sink_status);
     return;
   }
 
   output_device_id_ = device_id;
   security_origin_ = security_origin;
   bool was_sink_started = sink_started_;
 
   if (sink_.get())
     sink_->Stop();
 
   sink_started_ = false;
   sink_ = new_sink;
   if (was_sink_started)
     MaybeStartSink();
 
   callback.Run(media::OUTPUT_DEVICE_STATUS_OK);
 }
 
 media::AudioParameters TrackAudioRenderer::GetOutputParameters() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   if (!sink_ || !source_params_.IsValid())
     return media::AudioParameters();
 
   // Output parameters consist of the same channel layout and sample rate as the
   // source, but having the buffer duration preferred by the hardware.
-  const media::AudioParameters& preferred_params = sink_->GetOutputParameters();
+  const media::AudioParameters& preferred_params =
+      sink_->GetOutputDevice()->GetOutputParameters();
   return media::AudioParameters(
       preferred_params.format(), source_params_.channel_layout(),
       source_params_.sample_rate(), source_params_.bits_per_sample(),
       preferred_params.frames_per_buffer() * source_params_.sample_rate() /
           preferred_params.sample_rate());
 }
 
 media::OutputDeviceStatus TrackAudioRenderer::GetDeviceStatus() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   if (!sink_.get())
     return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
 
-  return sink_->GetDeviceStatus();
+  return sink_->GetOutputDevice()->GetDeviceStatus();
 }
 
 void TrackAudioRenderer::MaybeStartSink() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   DVLOG(1) << "TrackAudioRenderer::MaybeStartSink()";
 
   if (!sink_.get() || !source_params_.IsValid() || !playing_)
     return;
 
   // Re-create the AudioShifter to drop old audio data and reset to a starting
   // state.  MaybeStartSink() is always called in a situation where either the
   // source or sink has changed somehow and so all of AudioShifter's internal
   // time-sync state is invalid.
   CreateAudioShifter();
 
   if (sink_started_ ||
-      sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) {
+      sink_->GetOutputDevice()->GetDeviceStatus() !=
+          media::OUTPUT_DEVICE_STATUS_OK) {
     return;
   }
 
   DVLOG(1) << ("TrackAudioRenderer::MaybeStartSink() -- Starting sink.  "
                "source_params_={")
            << source_params_.AsHumanReadableString() << "}, sink parameters={"
            << GetOutputParameters().AsHumanReadableString() << '}';
   sink_->Initialize(GetOutputParameters(), this);
   sink_->Start();
   sink_->SetVolume(volume_);
+  sink_->Play();  // Not all the sinks play on start.
   sink_started_ = true;
   if (IsLocalRenderer()) {
     UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", kSinkStarted,
                               kSinkStatesMax);
   }
 }
 
 void TrackAudioRenderer::ReconfigureSink(const media::AudioParameters& params) {
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   DVLOG(1) << "TrackAudioRenderer::ReconfigureSink()";
 
   if (source_params_.Equals(params))
     return;
   source_params_ = params;
 
   if (!sink_.get())
     return;  // TrackAudioRenderer has not yet been started.
 
   // Stop |sink_| and re-create a new one to be initialized with different audio
   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
   sink_->Stop();
   sink_started_ = false;
-  sink_ =
-      AudioDeviceFactory::NewOutputDevice(playout_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
+  sink_ = AudioDeviceFactory::NewAudioRendererSink(
+      AudioDeviceFactory::kSourceNonRtcAudioTrack, playout_render_frame_id_,
+      session_id_, output_device_id_, security_origin_);
   MaybeStartSink();
 }
 
 void TrackAudioRenderer::CreateAudioShifter() {
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   // Note 1: The max buffer is fairly large to cover the case where
   // remotely-sourced audio is delivered well ahead of its scheduled playout
   // time (e.g., content streaming with a very large end-to-end
   // latency). However, there is no penalty for making it large in the
@@ skipping 20 matching lines @@
   if (source_params_.IsValid()) {
     prior_elapsed_render_time_ =
         ComputeTotalElapsedRenderTime(prior_elapsed_render_time_,
                                       num_samples_rendered_,
                                       source_params_.sample_rate());
     num_samples_rendered_ = 0;
   }
 }
 
 }  // namespace content