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

content/renderer/media/webrtc/processed_local_audio_source.cc

Index: content/renderer/media/webrtc/processed_local_audio_source.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc/processed_local_audio_source.cc
similarity index 23%
rename from content/renderer/media/webrtc_audio_capturer.cc
rename to content/renderer/media/webrtc/processed_local_audio_source.cc
index 113233ebefd60e3db1ac9a3c08c5e04fc6d18349..7d4ab030965ac529b4a0425461b90b50fd526f05 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc/processed_local_audio_source.cc
@@ -1,35 +1,79 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Copyright 2016 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_audio_capturer.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
 
-#include "base/bind.h"
 #include "base/logging.h"
-#include "base/macros.h"
 #include "base/metrics/histogram.h"
-#include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
-#include "build/build_config.h"
-#include "content/child/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
-#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_constraints_util.h"
+#include "content/renderer/media/rtc_media_constraints.h"
+#include "content/renderer/media/webrtc/processed_local_audio_track.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/media/webrtc_logging.h"
+#include "content/renderer/render_frame_impl.h"
+#include "media/audio/audio_input_device.h"
 #include "media/audio/sample_rates.h"
+#include "media/base/channel_layout.h"
+#include "third_party/webrtc/api/mediaconstraintsinterface.h"
 
 namespace content {
 
 namespace {
 
-// Audio buffer sizes are specified in milliseconds.
-const char kAudioLatency[] = "latencyMs";
-const int kMinAudioLatencyMs = 0;
-const int kMaxAudioLatencyMs = 10000;
+// Used as an identifier for ProcessedLocalAudioSource::From().
+void* const kClassIdentifier = const_cast<void**>(&kClassIdentifier);
+
+// Map of corresponding media constraints and platform effects.
+struct {
+  const char* constraint;
+  const media::AudioParameters::PlatformEffectsMask effect;
+} const kConstraintEffectMap[] = {
+  { webrtc::MediaConstraintsInterface::kGoogEchoCancellation,
+    media::AudioParameters::ECHO_CANCELLER },
+};
+
+// If any platform effects are available, check them against the constraints.
+// Disable effects to match false constraints, but if a constraint is true, set
+// the constraint to false to later disable the software effect.
+//
+// This function may modify both |constraints| and |effects|.
+void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
+                                    int* effects) {
+  if (*effects != media::AudioParameters::NO_EFFECTS) {
+    for (size_t i = 0; i < arraysize(kConstraintEffectMap); ++i) {
+      bool value;
+      size_t is_mandatory = 0;
+      if (!webrtc::FindConstraint(constraints,
+                                  kConstraintEffectMap[i].constraint,
+                                  &value,
+                                  &is_mandatory) || !value) {
+        // If the constraint is false, or does not exist, disable the platform
+        // effect.
+        *effects &= ~kConstraintEffectMap[i].effect;
+        DVLOG(1) << "Disabling platform effect: "
+                 << kConstraintEffectMap[i].effect;
+      } else if (*effects & kConstraintEffectMap[i].effect) {
+        // If the constraint is true, leave the platform effect enabled, and
+        // set the constraint to false to later disable the software effect.
+        if (is_mandatory) {
+          constraints->AddMandatory(kConstraintEffectMap[i].constraint,
+              webrtc::MediaConstraintsInterface::kValueFalse, true);
+        } else {
+          constraints->AddOptional(kConstraintEffectMap[i].constraint,
+              webrtc::MediaConstraintsInterface::kValueFalse, true);
+        }
+        DVLOG(1) << "Disabling constraint: "
+                 << kConstraintEffectMap[i].constraint;
+      }
+    }
+  }
+}
 
 // Method to check if any of the data in |audio_source| has energy.
 bool HasDataEnergy(const media::AudioBus& audio_source) {
@@ -47,129 +91,157 @@ bool HasDataEnergy(const media::AudioBus& audio_source) {
 
 }  // namespace
 
-// Reference counted container of WebRtcLocalAudioTrack delegate.
-// TODO(xians): Switch to MediaStreamAudioSinkOwner.
-class WebRtcAudioCapturer::TrackOwner
-    : public base::RefCountedThreadSafe<WebRtcAudioCapturer::TrackOwner> {
- public:
-  explicit TrackOwner(WebRtcLocalAudioTrack* track)
-      : delegate_(track) {}
-
-  void Capture(const media::AudioBus& audio_bus,
-               base::TimeTicks estimated_capture_time,
-               bool force_report_nonzero_energy) {
-    base::AutoLock lock(lock_);
-    if (delegate_) {
-      delegate_->Capture(audio_bus,
-                         estimated_capture_time,
-                         force_report_nonzero_energy);
-    }
-  }
+ProcessedLocalAudioSource::ProcessedLocalAudioSource(
+    int consumer_render_frame_id,
+    const StreamDeviceInfo& device_info,
+    PeerConnectionDependencyFactory* factory)
+    : MediaStreamAudioSource(true /* is_local_source */),
+      consumer_render_frame_id_(consumer_render_frame_id),
+      pc_factory_(factory),
+      exposed_volume_(0),
+      allow_invalid_render_frame_id_for_testing_(false) {
+  DCHECK(pc_factory_);
+  DVLOG(1) << "ProcessedLocalAudioSource::ProcessedLocalAudioSource()";
+  MediaStreamSource::SetDeviceInfo(device_info);
+}
 
-  void OnSetFormat(const media::AudioParameters& params) {
-    base::AutoLock lock(lock_);
-    if (delegate_)
-      delegate_->OnSetFormat(params);
-  }
+ProcessedLocalAudioSource::~ProcessedLocalAudioSource() {
+  DVLOG(1) << "ProcessedLocalAudioSource::~ProcessedLocalAudioSource()";
+  // Superclass will call StopSource() just in case.
+}
 
-  void SetAudioProcessor(
-      const scoped_refptr<MediaStreamAudioProcessor>& processor) {
-    base::AutoLock lock(lock_);
-    if (delegate_)
-      delegate_->SetAudioProcessor(processor);
-  }
+// static
+ProcessedLocalAudioSource* ProcessedLocalAudioSource::From(
+    MediaStreamAudioSource* source) {
+  if (source && source->GetClassIdentifier() == kClassIdentifier)
+    return static_cast<ProcessedLocalAudioSource*>(source);
+  return nullptr;
+}
 
-  void Reset() {
-    base::AutoLock lock(lock_);
-    delegate_ = NULL;
-  }
+void ProcessedLocalAudioSource::SetSourceConstraints(
+    const blink::WebMediaConstraints& constraints) {
+  DCHECK(!constraints.isNull());
+  constraints_ = constraints;
+}
 
-  void Stop() {
-    base::AutoLock lock(lock_);
-    DCHECK(delegate_);
+void* ProcessedLocalAudioSource::GetClassIdentifier() const {
+  return kClassIdentifier;
+}
 
-    // This can be reentrant so reset |delegate_| before calling out.
-    WebRtcLocalAudioTrack* temp = delegate_;
-    delegate_ = NULL;
-    temp->Stop();
+void ProcessedLocalAudioSource::DoStopSource() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (is_stopped_)
+    return;
+
+  // Setting |is_stopped_| while holding the |volume_lock_| because the
+  // SetVolume() method needs to know whether |input_device_| is valid.
+  {
+    base::AutoLock auto_lock(volume_lock_);
+    is_stopped_ = true;
   }
 
-  // Wrapper which allows to use std::find_if() when adding and removing
-  // sinks to/from the list.
-  struct TrackWrapper {
-    explicit TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {}
-    bool operator()(
-        const scoped_refptr<WebRtcAudioCapturer::TrackOwner>& owner) const {
-      return owner->IsEqual(track_);
+  if (input_device_) {
+    if (WebRtcAudioDeviceImpl* rtc_audio_device =
+            pc_factory_->GetWebRtcAudioDevice()) {
+      rtc_audio_device->RemoveAudioCapturer(this);
     }
-    WebRtcLocalAudioTrack* track_;
-  };
 
- protected:
-  virtual ~TrackOwner() {}
+    input_device_->Stop();
 
- private:
-  friend class base::RefCountedThreadSafe<WebRtcAudioCapturer::TrackOwner>;
+    // Stop the audio processor to avoid feeding render data into the processor.
+    audio_processor_->Stop();
 
-  bool IsEqual(const WebRtcLocalAudioTrack* other) const {
-    base::AutoLock lock(lock_);
-    return (other == delegate_);
+    VLOG(1) << "Stopped WebRTC audio pipeline for consumption by render frame "
+            << consumer_render_frame_id_ << '.';
   }
+}
 
-  // Do NOT reference count the |delegate_| to avoid cyclic reference counting.
-  WebRtcLocalAudioTrack* delegate_;
-  mutable base::Lock lock_;
-
-  DISALLOW_COPY_AND_ASSIGN(TrackOwner);
-};
+scoped_ptr<MediaStreamAudioTrack>
+ProcessedLocalAudioSource::CreateMediaStreamAudioTrack(
+    const std::string& id) {
+  DCHECK(thread_checker_.CalledOnValidThread());
 
-// static
-scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer(
-    int render_frame_id,
-    const StreamDeviceInfo& device_info,
-    const blink::WebMediaConstraints& constraints,
-    WebRtcAudioDeviceImpl* audio_device,
-    MediaStreamAudioSource* audio_source) {
-  scoped_refptr<WebRtcAudioCapturer> capturer = new WebRtcAudioCapturer(
-      render_frame_id, device_info, constraints, audio_device, audio_source);
-  if (capturer->Initialize())
-    return capturer;
-
-  return NULL;
+  ProcessedLocalAudioTrack* const audio_track =
+      new ProcessedLocalAudioTrack(WebRtcLocalAudioTrackAdapter::Create(
+          id, rtc_source_.get(), pc_factory_->GetWebRtcSignalingThread()));
+  audio_track->adapter()->SetAudioProcessor(audio_processor_);
+  audio_track->adapter()->SetReportedLevel(level_calculator_.reported_level());
+  return scoped_ptr<MediaStreamAudioTrack>(audio_track);
 }
 
-bool WebRtcAudioCapturer::Initialize() {
+bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
   DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
-  WebRtcLogMessage(base::StringPrintf(
-      "WAC::Initialize. render_frame_id=%d"
-      ", channel_layout=%d, sample_rate=%d, buffer_size=%d"
-      ", session_id=%d, paired_output_sample_rate=%d"
-      ", paired_output_frames_per_buffer=%d, effects=%d. ",
-      render_frame_id_, device_info_.device.input.channel_layout,
-      device_info_.device.input.sample_rate,
-      device_info_.device.input.frames_per_buffer, device_info_.session_id,
-      device_info_.device.matched_output.sample_rate,
-      device_info_.device.matched_output.frames_per_buffer,
-      device_info_.device.input.effects));
-
-  if (render_frame_id_ == -1) {
-    // Return true here to allow injecting a new source via
-    // SetCapturerSourceForTesting() at a later state.
+
+  if (is_stopped_)
+    return false;
+  if (input_device_)
     return true;
+
+  // Sanity-check that the consuming RenderFrame still exists.  This is
+  // required to initialize the audio source.
+  if (!allow_invalid_render_frame_id_for_testing_ &&
+      !RenderFrameImpl::FromRoutingID(consumer_render_frame_id_)) {
+    WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
+                     " because the render frame does not exist.");
+    StopSource();
+    return false;
   }
 
-  MediaAudioConstraints audio_constraints(constraints_,
-                                          device_info_.device.input.effects);
-  if (!audio_constraints.IsValid())
+  // Using |constraints_| as a basis, apply additional default constraints for
+  // audio processing and take the |effects| from StreamDeviceInfo into account.
+  //
+  // TODO(miu): Consolidation of logic needed here: There is both a
+  // RTCMediaConstraints and MediaAudioConstraints class, plus the constraints
+  // are being modified both within and outside this module.  (This problem was
+  // exposed after a major refactoring.)
+  RTCMediaConstraints rtc_constraints(constraints_);
+  MediaAudioConstraints::ApplyFixedAudioConstraints(&rtc_constraints);
+  StreamDeviceInfo modified_device_info = device_info();
+  HarmonizeConstraintsAndEffects(&rtc_constraints,
+                                 &modified_device_info.device.input.effects);
+  MediaStreamSource::SetDeviceInfo(modified_device_info);
+  MediaAudioConstraints audio_constraints(
+      constraints_, modified_device_info.device.input.effects);
+  if (!audio_constraints.IsValid()) {
+    WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
+                     " because MediaAudioConstraints are not valid.");
+    StopSource();
     return false;
+  }
 
-  media::ChannelLayout channel_layout = static_cast<media::ChannelLayout>(
-      device_info_.device.input.channel_layout);
+  const MediaStreamDevice::AudioDeviceParameters& source_params =
+      modified_device_info.device.input;
+  const MediaStreamDevice::AudioDeviceParameters& matched_params =
+      modified_device_info.device.matched_output;
+  WebRtcLogMessage(base::StringPrintf(
+      "ProcessedLocalAudioSource::EnsureSourceIsStarted.  PRELIMINARY "
+      "parameters: render_frame_id=%d"
+      ", channel_layout=%d, sample_rate=%d, buffer_size=%d"
+      ", session_id=%d, paired_output_sample_rate=%d"
+      ", paired_output_frames_per_buffer=%d, effects=%d.",
+      consumer_render_frame_id_, source_params.channel_layout,
+      source_params.sample_rate, source_params.frames_per_buffer,
+      modified_device_info.session_id, matched_params.sample_rate,
+      matched_params.frames_per_buffer, source_params.effects));
+
+  // Create the MediaStreamAudioProcessor, bound to the WebRTC audio device
+  // module.
+  WebRtcAudioDeviceImpl* const rtc_audio_device =
+      pc_factory_->GetWebRtcAudioDevice();
+  if (!rtc_audio_device) {
+    WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
+                     " because there is no WebRtcAudioDeviceImpl instance.");
+    StopSource();
+    return false;
+  }
+  audio_processor_ = new rtc::RefCountedObject<MediaStreamAudioProcessor>(
+      constraints_, source_params, rtc_audio_device);
 
   // If KEYBOARD_MIC effect is set, change the layout to the corresponding
   // layout that includes the keyboard mic.
-  if ((device_info_.device.input.effects &
+  media::ChannelLayout channel_layout = static_cast<media::ChannelLayout>(
+      modified_device_info.device.input.channel_layout);
+  if ((modified_device_info.device.input.effects &
           media::AudioParameters::KEYBOARD_MIC) &&
       audio_constraints.GetProperty(
           MediaAudioConstraints::kGoogExperimentalNoiseSuppression)) {
@@ -182,299 +254,117 @@ bool WebRtcAudioCapturer::Initialize() {
                << channel_layout;
     }
   }
-
   DVLOG(1) << "Audio input hardware channel layout: " << channel_layout;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX + 1);
 
-  // Verify that the reported input channel configuration is supported.
+  // Verify that the input channel configuration is supported.
   if (channel_layout != media::CHANNEL_LAYOUT_MONO &&
       channel_layout != media::CHANNEL_LAYOUT_STEREO &&
       channel_layout != media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC) {
-    DLOG(ERROR) << channel_layout
-                << " is not a supported input channel configuration.";
+    WebRtcLogMessage(base::StringPrintf(
+        "ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
+        " because the input channel layout (%d) is not supported.",
+        static_cast<int>(channel_layout)));
+    StopSource();
     return false;
   }
 
   DVLOG(1) << "Audio input hardware sample rate: "
-           << device_info_.device.input.sample_rate;
+           << modified_device_info.device.input.sample_rate;
   media::AudioSampleRate asr;
-  if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) {
+  if (media::ToAudioSampleRate(modified_device_info.device.input.sample_rate,
+                               &asr)) {
     UMA_HISTOGRAM_ENUMERATION(
         "WebRTC.AudioInputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected",
-                         device_info_.device.input.sample_rate);
-  }
-
-  // Initialize the buffer size to zero, which means it wasn't specified.
-  // If it is out of range, we return it to zero.
-  int buffer_size_ms = 0;
-  int buffer_size_samples = 0;
-  GetConstraintValueAsInteger(constraints_, kAudioLatency, &buffer_size_ms);
-  if (buffer_size_ms < kMinAudioLatencyMs ||
-      buffer_size_ms > kMaxAudioLatencyMs) {
-    DVLOG(1) << "Ignoring out of range buffer size " << buffer_size_ms;
-  } else {
-    buffer_size_samples =
-        device_info_.device.input.sample_rate * buffer_size_ms / 1000;
-  }
-  DVLOG_IF(1, buffer_size_samples > 0)
-      << "Custom audio buffer size: " << buffer_size_samples << " samples";
-
-  // Create and configure the default audio capturing source.
-  SetCapturerSourceInternal(
-      AudioDeviceFactory::NewInputDevice(render_frame_id_),
-      channel_layout,
-      device_info_.device.input.sample_rate,
-      buffer_size_samples);
-
-  // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
-  // information from the capturer.
-  if (audio_device_)
-    audio_device_->AddAudioCapturer(this);
-
-  return true;
-}
-
-WebRtcAudioCapturer::WebRtcAudioCapturer(
-    int render_frame_id,
-    const StreamDeviceInfo& device_info,
-    const blink::WebMediaConstraints& constraints,
-    WebRtcAudioDeviceImpl* audio_device,
-    MediaStreamAudioSource* audio_source)
-    : constraints_(constraints),
-      audio_processor_(new rtc::RefCountedObject<MediaStreamAudioProcessor>(
-          constraints,
-          device_info.device.input,
-          audio_device)),
-      running_(false),
-      render_frame_id_(render_frame_id),
-      device_info_(device_info),
-      volume_(0),
-      peer_connection_mode_(false),
-      audio_device_(audio_device),
-      audio_source_(audio_source) {
-  DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
-}
-
-WebRtcAudioCapturer::~WebRtcAudioCapturer() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(tracks_.IsEmpty());
-  DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()";
-  Stop();
-}
-
-void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) {
-  DCHECK(track);
-  DVLOG(1) << "WebRtcAudioCapturer::AddTrack()";
-
-  {
-    base::AutoLock auto_lock(lock_);
-    // Verify that |track| is not already added to the list.
-    DCHECK(!tracks_.Contains(TrackOwner::TrackWrapper(track)));
-
-    // Add with a tag, so we remember to call OnSetFormat() on the new
-    // track.
-    scoped_refptr<TrackOwner> track_owner(new TrackOwner(track));
-    tracks_.AddAndTag(track_owner.get());
-  }
-}
-
-void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcAudioCapturer::RemoveTrack()";
-  bool stop_source = false;
-  {
-    base::AutoLock auto_lock(lock_);
-
-    scoped_refptr<TrackOwner> removed_item =
-        tracks_.Remove(TrackOwner::TrackWrapper(track));
-
-    // Clear the delegate to ensure that no more capture callbacks will
-    // be sent to this sink. Also avoids a possible crash which can happen
-    // if this method is called while capturing is active.
-    if (removed_item.get()) {
-      removed_item->Reset();
-      stop_source = tracks_.IsEmpty();
-    }
-  }
-  if (stop_source) {
-    // Since WebRtcAudioCapturer does not inherit MediaStreamAudioSource,
-    // and instead MediaStreamAudioSource is composed of a WebRtcAudioCapturer,
-    // we have to call StopSource on the MediaStreamSource. This will call
-    // MediaStreamAudioSource::DoStopSource which in turn call
-    // WebRtcAudioCapturerer::Stop();
-    audio_source_->StopSource();
+                         modified_device_info.device.input.sample_rate);
   }
-}
 
-void WebRtcAudioCapturer::SetCapturerSourceInternal(
-    const scoped_refptr<media::AudioCapturerSource>& source,
-    media::ChannelLayout channel_layout,
-    int sample_rate,
-    int buffer_size) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
-           << "sample_rate=" << sample_rate << ")";
-  scoped_refptr<media::AudioCapturerSource> old_source;
-  {
-    base::AutoLock auto_lock(lock_);
-    if (source_.get() == source.get())
-      return;
-
-    source_.swap(old_source);
-    source_ = source;
-
-    // Reset the flag to allow starting the new source.
-    running_ = false;
-  }
-
-  DVLOG(1) << "Switching to a new capture source.";
-  if (old_source.get())
-    old_source->Stop();
-
-  // If the buffer size is zero, it has not been specified.
-  // We either default to 10ms, or use the hardware buffer size.
-  if (buffer_size == 0)
-    buffer_size = GetBufferSize(sample_rate);
-
-  // Dispatch the new parameters both to the sink(s) and to the new source,
-  // also apply the new |constraints|.
-  // The idea is to get rid of any dependency of the microphone parameters
-  // which would normally be used by default.
-  // bits_per_sample is always 16 for now.
-  media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                                channel_layout, sample_rate, 16, buffer_size);
-  params.set_effects(device_info_.device.input.effects);
-
-  {
-    base::AutoLock auto_lock(lock_);
-    // Notify the |audio_processor_| of the new format.
-    audio_processor_->OnCaptureFormatChanged(params);
-
-    // Notify all tracks about the new format.
-    tracks_.TagAll();
-  }
-
-  if (source.get())
-    source->Initialize(params, this, session_id());
-
-  Start();
-}
-
-void WebRtcAudioCapturer::EnablePeerConnectionMode() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "EnablePeerConnectionMode";
-  // Do nothing if the peer connection mode has been enabled.
-  if (peer_connection_mode_)
-    return;
-
-  peer_connection_mode_ = true;
-  int render_frame_id = -1;
-  media::AudioParameters input_params;
-  {
-    base::AutoLock auto_lock(lock_);
-    // Simply return if there is no existing source or the |render_frame_id_| is
-    // not valid.
-    if (!source_.get() || render_frame_id_ == -1)
-      return;
-
-    render_frame_id = render_frame_id_;
-    input_params = audio_processor_->InputFormat();
-  }
+  // The buffer size is 20 ms on Android, and 10 ms everywhere else.
+#if defined(OS_ANDROID)
+  // TODO(henrika): Tune and adjust buffer size on Android.
+  const int buffer_size_samples =
+      modified_device_info.device.input.sample_rate / 50;
+#else
+  const int buffer_size_samples =
+      modified_device_info.device.input.sample_rate / 100;
+#endif
 
-  // Do nothing if the current buffer size is the WebRtc native buffer size.
-  if (GetBufferSize(input_params.sample_rate()) ==
-          input_params.frames_per_buffer()) {
-    return;
+  // Determine the audio format required of the AudioInputDevice.  Then, pass
+  // that to the |audio_processor_| and set the output format of this
+  // ProcessedLocalAudioSource to the processor's output format.
+  media::AudioParameters params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
+      modified_device_info.device.input.sample_rate, 16, buffer_size_samples);
+  params.set_effects(modified_device_info.device.input.effects);
+  DCHECK(params.IsValid());
+  audio_processor_->OnCaptureFormatChanged(params);
+  MediaStreamAudioSource::SetFormat(audio_processor_->OutputFormat());
+
+  // Start the source.
+  VLOG(1) << "Starting WebRTC audio source for consumption by render frame "
+          << consumer_render_frame_id_ << " with audio parameters={"
+          << GetAudioParameters().AsHumanReadableString() << '}';
+  scoped_refptr<media::AudioInputDevice> device =
+      AudioDeviceFactory::NewInputDevice(consumer_render_frame_id_);
+  device->Initialize(params, this, modified_device_info.session_id);
+  // We need to set the AGC control before starting the stream.
+  device->SetAutomaticGainControl(true);
+  device->Start();
+  input_device_ = device;  // Thread-safe assignment.
+
+  // Register this source with the WebRtcAudioDeviceImpl.
+  rtc_audio_device->AddAudioCapturer(this);
+
+  // Creates a LocalAudioSource object which holds audio options.
+  // TODO(xians): The option should apply to the track instead of the source.
+  // TODO(perkj): Move audio constraints parsing to Chrome.
+  // Currently there are a few constraints that are parsed by libjingle and
+  // the state is set to ended if parsing fails.
+  rtc_source_ = pc_factory_->CreateLocalAudioSource(&rtc_constraints);
+  if (rtc_source_->state() != webrtc::MediaSourceInterface::kLive) {
+    WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
+                     " because the rtc LocalAudioSource is not live.");
+    StopSource();
+    return false;
   }
 
-  // Create a new audio stream as source which will open the hardware using
-  // WebRtc native buffer size.
-  SetCapturerSourceInternal(AudioDeviceFactory::NewInputDevice(render_frame_id),
-                            input_params.channel_layout(),
-                            input_params.sample_rate(),
-                            0);
+  return true;
 }
 
-void WebRtcAudioCapturer::Start() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcAudioCapturer::Start()";
-  base::AutoLock auto_lock(lock_);
-  if (running_ || !source_.get())
-    return;
-
-  // Start the data source, i.e., start capturing data from the current source.
-  // We need to set the AGC control before starting the stream.
-  source_->SetAutomaticGainControl(true);
-  source_->Start();
-  running_ = true;
-}
+void ProcessedLocalAudioSource::SetVolume(int volume) {
+  DVLOG(1) << "ProcessedLocalAudioSource::SetVolume()";
+  DCHECK_LE(volume, MaxVolume());
 
-void WebRtcAudioCapturer::Stop() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcAudioCapturer::Stop()";
-  scoped_refptr<media::AudioCapturerSource> source;
-  TrackList::ItemList tracks;
   {
-    base::AutoLock auto_lock(lock_);
-    if (!running_)
+    base::AutoLock auto_lock(volume_lock_);
+    if (is_stopped_)
       return;
-
-    source = source_;
-    tracks = tracks_.Items();
-    tracks_.Clear();
-    running_ = false;
   }
 
-  // Remove the capturer object from the WebRtcAudioDeviceImpl.
-  if (audio_device_)
-    audio_device_->RemoveAudioCapturer(this);
-
-  for (TrackList::ItemList::const_iterator it = tracks.begin();
-       it != tracks.end();
-       ++it) {
-    (*it)->Stop();
+  // Assumption: Once |input_device_| is set, it will never change.  Thus,
+  // there's no need to hold any locks for the following:
+  if (input_device_.get()) {
+    double normalized_volume = static_cast<double>(volume) / MaxVolume();
+    input_device_->SetVolume(normalized_volume);
   }
-
-  if (source.get())
-    source->Stop();
-
-  // Stop the audio processor to avoid feeding render data into the processor.
-  audio_processor_->Stop();
 }
 
-void WebRtcAudioCapturer::SetVolume(int volume) {
-  DVLOG(1) << "WebRtcAudioCapturer::SetVolume()";
-  DCHECK_LE(volume, MaxVolume());
-  double normalized_volume = static_cast<double>(volume) / MaxVolume();
-  base::AutoLock auto_lock(lock_);
-  if (source_.get())
-    source_->SetVolume(normalized_volume);
-}
-
-int WebRtcAudioCapturer::Volume() const {
-  base::AutoLock auto_lock(lock_);
-  return volume_;
+int ProcessedLocalAudioSource::Volume() const {
+  base::AutoLock auto_lock(volume_lock_);
+  return exposed_volume_;
 }
 
-int WebRtcAudioCapturer::MaxVolume() const {
+int ProcessedLocalAudioSource::MaxVolume() const {
   return WebRtcAudioDeviceImpl::kMaxVolumeLevel;
 }
 
-media::AudioParameters WebRtcAudioCapturer::GetOutputFormat() const {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  return audio_processor_->OutputFormat();
-}
-
-void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
-                                  int audio_delay_milliseconds,
-                                  double volume,
-                                  bool key_pressed) {
-// This callback is driven by AudioInputDevice::AudioThreadCallback if
-// |source_| is AudioInputDevice, otherwise it is driven by client's
-// CaptureCallback.
+void ProcessedLocalAudioSource::Capture(const media::AudioBus* audio_bus,
+                                        int audio_delay_milliseconds,
+                                        double volume,
+                                        bool key_pressed) {
 #if defined(OS_WIN) || defined(OS_MACOSX)
   DCHECK_LE(volume, 1.0);
 #elif (defined(OS_LINUX) && !defined(OS_CHROMEOS)) || defined(OS_OPENBSD)
@@ -491,47 +381,27 @@ void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
   // audio/video sync.  http://crbug.com/335335
   const base::TimeTicks reference_clock_snapshot = base::TimeTicks::Now();
 
-  TrackList::ItemList tracks;
-  TrackList::ItemList tracks_to_notify_format;
-  int current_volume = 0;
+  // Map internal volume range of [0.0, 1.0] into [0, 255] used by AGC.
+  int current_volume = static_cast<int>((volume * MaxVolume()) + 0.5);
   {
-    base::AutoLock auto_lock(lock_);
-    if (!running_)
-      return;
-
-    // Map internal volume range of [0.0, 1.0] into [0, 255] used by AGC.
-    // The volume can be higher than 255 on Linux, and it will be cropped to
-    // 255 since AGC does not allow values out of range.
-    volume_ = static_cast<int>((volume * MaxVolume()) + 0.5);
-    current_volume = volume_ > MaxVolume() ? MaxVolume() : volume_;
-    tracks = tracks_.Items();
-    tracks_.RetrieveAndClearTags(&tracks_to_notify_format);
+    base::AutoLock auto_lock(volume_lock_);
+    exposed_volume_ = current_volume;
   }
+  // The volume can be higher than 255 on Linux, and it will be cropped to
+  // 255 since AGC does not allow values out of range.
+  current_volume = current_volume > MaxVolume() ? MaxVolume() : current_volume;
 
   DCHECK(audio_processor_->InputFormat().IsValid());
-  DCHECK_EQ(audio_source->channels(),
-            audio_processor_->InputFormat().channels());
-  DCHECK_EQ(audio_source->frames(),
+  DCHECK_EQ(audio_bus->channels(), audio_processor_->InputFormat().channels());
+  DCHECK_EQ(audio_bus->frames(),
             audio_processor_->InputFormat().frames_per_buffer());
 
-  // Notify the tracks on when the format changes. This will do nothing if
-  // |tracks_to_notify_format| is empty.
-  const media::AudioParameters& output_params =
-      audio_processor_->OutputFormat();
-  for (const auto& track : tracks_to_notify_format) {
-    track->OnSetFormat(output_params);
-    track->SetAudioProcessor(audio_processor_);
-  }
-
-  // Figure out if the pre-processed data has any energy or not, the
-  // information will be passed to the track to force the calculator
-  // to report energy in case the post-processed data is zeroed by the audio
-  // processing.
-  const bool force_report_nonzero_energy = HasDataEnergy(*audio_source);
+  // Figure out if the pre-processed data has any energy or not.
+  const bool force_report_nonzero_energy = HasDataEnergy(*audio_bus);
 
   // Push the data to the processor for processing.
   audio_processor_->PushCaptureData(
-      *audio_source,
+      *audio_bus,
       base::TimeDelta::FromMilliseconds(audio_delay_milliseconds));
 
   // Process and consume the data in the processor until there is not enough
@@ -543,85 +413,30 @@ void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
              current_volume, key_pressed,
              &processed_data, &processed_data_audio_delay, &new_volume)) {
     DCHECK(processed_data);
-    const base::TimeTicks processed_data_capture_time =
-        reference_clock_snapshot - processed_data_audio_delay;
-    for (const auto& track : tracks) {
-      track->Capture(*processed_data,
-                     processed_data_capture_time,
-                     force_report_nonzero_energy);
-    }
 
-    if (new_volume) {
-      SetVolume(new_volume);
+    level_calculator_.Calculate(*processed_data, force_report_nonzero_energy);
+
+    MediaStreamAudioSource::DeliverDataToTracks(
+        *processed_data, reference_clock_snapshot - processed_data_audio_delay);
 
-      // Update the |current_volume| to avoid passing the old volume to AGC.
+    // TODO(xians): This could result in an IPC call being made for each audio
+    // chunk (!).  Consider adding throttling logic here.
+    if (new_volume != current_volume) {
+      SetVolume(new_volume);
       current_volume = new_volume;
     }
   }
 }
 
-void WebRtcAudioCapturer::OnCaptureError(const std::string& message) {
-  WebRtcLogMessage("WAC::OnCaptureError: " + message);
-}
-
-media::AudioParameters WebRtcAudioCapturer::source_audio_parameters() const {
-  base::AutoLock auto_lock(lock_);
-  return audio_processor_.get() ? audio_processor_->InputFormat()
-                                : media::AudioParameters();
-}
-
-bool WebRtcAudioCapturer::GetPairedOutputParameters(
-    int* session_id,
-    int* output_sample_rate,
-    int* output_frames_per_buffer) const {
-  // Don't set output parameters unless all of them are valid.
-  if (device_info_.session_id <= 0 ||
-      !device_info_.device.matched_output.sample_rate ||
-      !device_info_.device.matched_output.frames_per_buffer)
-    return false;
-
-  *session_id = device_info_.session_id;
-  *output_sample_rate = device_info_.device.matched_output.sample_rate;
-  *output_frames_per_buffer =
-      device_info_.device.matched_output.frames_per_buffer;
-
-  return true;
-}
-
-int WebRtcAudioCapturer::GetBufferSize(int sample_rate) const {
-  DCHECK(thread_checker_.CalledOnValidThread());
-#if defined(OS_ANDROID)
-  // TODO(henrika): Tune and adjust buffer size on Android.
-  return (2 * sample_rate / 100);
-#endif
-
-  // PeerConnection is running at a buffer size of 10ms data. A multiple of
-  // 10ms as the buffer size can give the best performance to PeerConnection.
-  int peer_connection_buffer_size = sample_rate / 100;
-
-  // Use the native hardware buffer size in non peer connection mode when the
-  // platform is using a native buffer size smaller than the PeerConnection
-  // buffer size and audio processing is off.
-  int hardware_buffer_size = device_info_.device.input.frames_per_buffer;
-  if (!peer_connection_mode_ && hardware_buffer_size &&
-      hardware_buffer_size <= peer_connection_buffer_size &&
-      !audio_processor_->has_audio_processing()) {
-    DVLOG(1) << "WebRtcAudioCapturer is using hardware buffer size "
-             << hardware_buffer_size;
-    return hardware_buffer_size;
-  }
-
-  return (sample_rate / 100);
+void ProcessedLocalAudioSource::OnCaptureError(const std::string& message) {
+  WebRtcLogMessage("ProcessedLocalAudioSource::OnCaptureError: " + message);
 }
 
-void WebRtcAudioCapturer::SetCapturerSource(
-    const scoped_refptr<media::AudioCapturerSource>& source,
-    media::AudioParameters params) {
-  // Create a new audio stream as source which uses the new source.
-  SetCapturerSourceInternal(source,
-                            params.channel_layout(),
-                            params.sample_rate(),
-                            0);
+media::AudioParameters ProcessedLocalAudioSource::GetInputAudioParameters()
+    const {
+  return audio_processor_.get()
+      ? audio_processor_->InputFormat()
+      : media::AudioParameters();
 }
 
 }  // namespace content