MediaStream audio rendering: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
deleted file mode 100644
index f39a017b94db441164f527fba3b8790aca8d46b0..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ /dev/null
@@ -1,356 +0,0 @@
-// 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/webrtc_local_audio_renderer.h"
-
-#include <utility>
-
-#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_dispatcher.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-#include "content/renderer/media/webrtc_audio_renderer.h"
-#include "content/renderer/render_frame_impl.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,
-  kSinkStatesMax  // Must always be last!
-};
-
-}  // namespace
-
-// media::AudioRendererSink::RenderCallback implementation
-int WebRtcLocalAudioRenderer::Render(media::AudioBus* audio_bus,
-                                     uint32_t audio_delay_milliseconds,
-                                     uint32_t frames_skipped) {
-  TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
-  base::AutoLock auto_lock(thread_lock_);
-
-  if (!playing_ || !volume_ || !audio_shifter_) {
-    audio_bus->Zero();
-    return 0;
-  }
-
-  audio_shifter_->Pull(
-      audio_bus,
-      base::TimeTicks::Now() -
-      base::TimeDelta::FromMilliseconds(audio_delay_milliseconds));
-
-  return audio_bus->frames();
-}
-
-void WebRtcLocalAudioRenderer::OnRenderError() {
-  NOTIMPLEMENTED();
-}
-
-// content::MediaStreamAudioSink implementation
-void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus,
-                                      base::TimeTicks estimated_capture_time) {
-  DCHECK(capture_thread_checker_.CalledOnValidThread());
-  DCHECK(!estimated_capture_time.is_null());
-
-  TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
-
-  base::AutoLock auto_lock(thread_lock_);
-  if (!playing_ || !volume_ || !audio_shifter_)
-    return;
-
-  scoped_ptr<media::AudioBus> audio_data(
-      media::AudioBus::Create(audio_bus.channels(), audio_bus.frames()));
-  audio_bus.CopyTo(audio_data.get());
-  audio_shifter_->Push(std::move(audio_data), estimated_capture_time);
-  const base::TimeTicks now = base::TimeTicks::Now();
-  total_render_time_ += now - last_render_time_;
-  last_render_time_ = now;
-}
-
-void WebRtcLocalAudioRenderer::OnSetFormat(
-    const media::AudioParameters& params) {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()";
-  // If the source is restarted, we might have changed to another capture
-  // thread.
-  capture_thread_checker_.DetachFromThread();
-  DCHECK(capture_thread_checker_.CalledOnValidThread());
-
-  // Post a task on the main render thread to reconfigure the |sink_| with the
-  // new format.
-  task_runner_->PostTask(
-      FROM_HERE,
-      base::Bind(&WebRtcLocalAudioRenderer::ReconfigureSink, this, params));
-}
-
-// WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer implementation.
-WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer(
-    const blink::WebMediaStreamTrack& audio_track,
-    int source_render_frame_id,
-    int session_id,
-    const std::string& device_id,
-    const url::Origin& security_origin)
-    : audio_track_(audio_track),
-      source_render_frame_id_(source_render_frame_id),
-      session_id_(session_id),
-      task_runner_(base::ThreadTaskRunnerHandle::Get()),
-      playing_(false),
-      output_device_id_(device_id),
-      security_origin_(security_origin),
-      volume_(0.0),
-      sink_started_(false) {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer()";
-}
-
-WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  DCHECK(!sink_.get());
-  DVLOG(1) << "WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer()";
-}
-
-void WebRtcLocalAudioRenderer::Start() {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::Start()";
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  // 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(source_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
-
-  base::AutoLock auto_lock(thread_lock_);
-  last_render_time_ = base::TimeTicks::Now();
-  playing_ = false;
-}
-
-void WebRtcLocalAudioRenderer::Stop() {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::Stop()";
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  {
-    base::AutoLock auto_lock(thread_lock_);
-    playing_ = false;
-    audio_shifter_.reset();
-  }
-
-  // Stop the output audio stream, i.e, stop asking for data to render.
-  // It is safer to call Stop() on the |sink_| to clean up the resources even
-  // when the |sink_| is never started.
-  if (sink_.get()) {
-    sink_->Stop();
-    sink_ = NULL;
-  }
-
-  if (!sink_started_) {
-    UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
-                              kSinkNeverStarted, kSinkStatesMax);
-  }
-  sink_started_ = false;
-
-  // Ensure that the capturer stops feeding us with captured audio.
-  MediaStreamAudioSink::RemoveFromAudioTrack(this, audio_track_);
-}
-
-void WebRtcLocalAudioRenderer::Play() {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::Play()";
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  if (!sink_.get())
-    return;
-
-  {
-    base::AutoLock auto_lock(thread_lock_);
-    // Resumes rendering by ensuring that WebRtcLocalAudioRenderer::Render()
-    // now reads data from the local FIFO.
-    playing_ = true;
-    last_render_time_ = base::TimeTicks::Now();
-  }
-
-  // Note: If volume_ is currently muted, the |sink_| will not be started yet.
-  MaybeStartSink();
-}
-
-void WebRtcLocalAudioRenderer::Pause() {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::Pause()";
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  if (!sink_.get())
-    return;
-
-  base::AutoLock auto_lock(thread_lock_);
-  // Temporarily suspends rendering audio.
-  // WebRtcLocalAudioRenderer::Render() will return early during this state
-  // and only zeros will be provided to the active sink.
-  playing_ = false;
-}
-
-void WebRtcLocalAudioRenderer::SetVolume(float volume) {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::SetVolume(" << volume << ")";
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  {
-    base::AutoLock auto_lock(thread_lock_);
-    // Cache the volume.
-    volume_ = volume;
-  }
-
-  // Lazily start the |sink_| when the local renderer is unmuted during
-  // playing.
-  MaybeStartSink();
-
-  if (sink_.get())
-    sink_->SetVolume(volume);
-}
-
-media::OutputDevice* WebRtcLocalAudioRenderer::GetOutputDevice() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  return this;
-}
-
-base::TimeDelta WebRtcLocalAudioRenderer::GetCurrentRenderTime() const {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  base::AutoLock auto_lock(thread_lock_);
-  if (!sink_.get())
-    return base::TimeDelta();
-  return total_render_time();
-}
-
-bool WebRtcLocalAudioRenderer::IsLocalRenderer() const {
-  return true;
-}
-
-void WebRtcLocalAudioRenderer::SwitchOutputDevice(
-    const std::string& device_id,
-    const url::Origin& security_origin,
-    const media::SwitchOutputDeviceCB& callback) {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::SwitchOutputDevice()";
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  scoped_refptr<media::AudioOutputDevice> new_sink =
-      AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_,
-                                          device_id, security_origin);
-  if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) {
-    callback.Run(new_sink->GetDeviceStatus());
-    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;
-  int frames_per_buffer = sink_->GetOutputParameters().frames_per_buffer();
-  sink_params_ = source_params_;
-  sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize(
-      source_params_.sample_rate(), frames_per_buffer));
-
-  if (was_sink_started)
-    MaybeStartSink();
-
-  callback.Run(media::OUTPUT_DEVICE_STATUS_OK);
-}
-
-media::AudioParameters WebRtcLocalAudioRenderer::GetOutputParameters() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  if (!sink_.get())
-    return media::AudioParameters();
-
-  return sink_->GetOutputParameters();
-}
-
-media::OutputDeviceStatus WebRtcLocalAudioRenderer::GetDeviceStatus() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  if (!sink_.get())
-    return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
-
-  return sink_->GetDeviceStatus();
-}
-
-void WebRtcLocalAudioRenderer::MaybeStartSink() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink()";
-
-  if (!sink_.get() || !source_params_.IsValid())
-    return;
-
-  {
-    // Clear up the old data in the FIFO.
-    base::AutoLock auto_lock(thread_lock_);
-    audio_shifter_->Flush();
-  }
-
-  if (!sink_params_.IsValid() || !playing_ || !volume_ || sink_started_ ||
-      sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK)
-    return;
-
-  DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_.";
-  sink_->Initialize(sink_params_, this);
-  sink_->Start();
-  sink_started_ = true;
-  UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
-                            kSinkStarted, kSinkStatesMax);
-}
-
-void WebRtcLocalAudioRenderer::ReconfigureSink(
-    const media::AudioParameters& params) {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-
-  DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()";
-
-  if (source_params_.Equals(params))
-    return;
-
-  // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
-  // the new format.
-
-  source_params_ = params;
-  {
-    // Note: The max buffer is fairly large, but will rarely be used.
-    // Cast needs the buffer to hold at least one second of audio.
-    // The clock accuracy is set to 20ms because clock accuracy is
-    // ~15ms on windows.
-    media::AudioShifter* const new_shifter = new media::AudioShifter(
-        base::TimeDelta::FromSeconds(2),
-        base::TimeDelta::FromMilliseconds(20),
-        base::TimeDelta::FromSeconds(20),
-        source_params_.sample_rate(),
-        params.channels());
-
-    base::AutoLock auto_lock(thread_lock_);
-    audio_shifter_.reset(new_shifter);
-  }
-
-  if (!sink_.get())
-    return;  // WebRtcLocalAudioRenderer 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(source_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
-  int frames_per_buffer = sink_->GetOutputParameters().frames_per_buffer();
-  sink_params_ = source_params_;
-  sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize(
-      source_params_.sample_rate(), frames_per_buffer));
-  MaybeStartSink();
-}
-
-}  // namespace content