gecko-dev/dom/media/webrtc/MediaEngineWebRTCAudio.cpp

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
 * License, v. 2.0. If a copy of the MPL was not distributed with this file,
 * You can obtain one at http://mozilla.org/MPL/2.0/. */

#include "MediaEngineWebRTCAudio.h"

#include <stdio.h>
#include <algorithm>

#include "AudioConverter.h"
#include "MediaManager.h"
#include "MediaStreamGraphImpl.h"
#include "MediaTrackConstraints.h"
#include "mozilla/Assertions.h"
#include "mozilla/ErrorNames.h"
#include "mtransport/runnable_utils.h"
#include "nsAutoPtr.h"
#include "Tracing.h"

// scoped_ptr.h uses FF
#ifdef FF
#undef FF
#endif
#include "webrtc/voice_engine/voice_engine_defines.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/common_audio/include/audio_util.h"

using namespace webrtc;

// These are restrictions from the webrtc.org code
#define MAX_CHANNELS 2
#define MAX_SAMPLING_FREQ 48000 // Hz - multiple of 100

#ifdef MOZ_PULSEAUDIO
static uint32_t sInputStreamsOpen = 0;
#endif

namespace mozilla {

#ifdef LOG
#undef LOG
#endif

LogModule*
GetMediaManagerLog();
#define LOG(msg) MOZ_LOG(GetMediaManagerLog(), mozilla::LogLevel::Debug, msg)
#define LOG_FRAMES(msg)                                                        \
  MOZ_LOG(GetMediaManagerLog(), mozilla::LogLevel::Verbose, msg)

/**
 * WebRTC Microphone MediaEngineSource.
 */

MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
  RefPtr<AudioDeviceInfo> aInfo,
  const nsString& aDeviceName,
  const nsCString& aDeviceUUID,
  uint32_t aMaxChannelCount,
  bool aDelayAgnostic,
  bool aExtendedFilter)
  : mTrackID(TRACK_NONE)
  , mPrincipal(PRINCIPAL_HANDLE_NONE)
  , mDeviceInfo(std::move(aInfo))
  , mDelayAgnostic(aDelayAgnostic)
  , mExtendedFilter(aExtendedFilter)
  , mDeviceName(aDeviceName)
  , mDeviceUUID(aDeviceUUID)
  , mDeviceMaxChannelCount(aMaxChannelCount)
  , mSettings(
      new nsMainThreadPtrHolder<media::Refcountable<dom::MediaTrackSettings>>(
        "MediaEngineWebRTCMicrophoneSource::mSettings",
        new media::Refcountable<dom::MediaTrackSettings>(),
        // Non-strict means it won't assert main thread for us.
        // It would be great if it did but we're already on the media thread.
        /* aStrict = */ false))
{
#ifndef ANDROID
  MOZ_ASSERT(mDeviceInfo->DeviceID());
#endif

  // We'll init lazily as needed
  mSettings->mEchoCancellation.Construct(0);
  mSettings->mAutoGainControl.Construct(0);
  mSettings->mNoiseSuppression.Construct(0);
  mSettings->mChannelCount.Construct(0);

  mState = kReleased;
}

nsString
MediaEngineWebRTCMicrophoneSource::GetName() const
{
  return mDeviceName;
}

nsCString
MediaEngineWebRTCMicrophoneSource::GetUUID() const
{
  return mDeviceUUID;
}

// GetBestFitnessDistance returns the best distance the capture device can offer
// as a whole, given an accumulated number of ConstraintSets.
// Ideal values are considered in the first ConstraintSet only.
// Plain values are treated as Ideal in the first ConstraintSet.
// Plain values are treated as Exact in subsequent ConstraintSets.
// Infinity = UINT32_MAX e.g. device cannot satisfy accumulated ConstraintSets.
// A finite result may be used to calculate this device's ranking as a choice.

uint32_t
MediaEngineWebRTCMicrophoneSource::GetBestFitnessDistance(
  const nsTArray<const NormalizedConstraintSet*>& aConstraintSets,
  const nsString& aDeviceId) const
{
  uint32_t distance = 0;

  for (const auto* cs : aConstraintSets) {
    distance =
      MediaConstraintsHelper::GetMinimumFitnessDistance(*cs, aDeviceId);
    break; // distance is read from first entry only
  }
  return distance;
}

nsresult
MediaEngineWebRTCMicrophoneSource::EvaluateSettings(
  const NormalizedConstraints& aConstraintsUpdate,
  const MediaEnginePrefs& aInPrefs,
  MediaEnginePrefs* aOutPrefs,
  const char** aOutBadConstraint)
{
  AssertIsOnOwningThread();

  MediaEnginePrefs prefs = aInPrefs;

  FlattenedConstraints c(aConstraintsUpdate);

  prefs.mAecOn = c.mEchoCancellation.Get(aInPrefs.mAecOn);
  prefs.mAgcOn = c.mAutoGainControl.Get(aInPrefs.mAgcOn);
  prefs.mNoiseOn = c.mNoiseSuppression.Get(aInPrefs.mNoiseOn);

  // Determine an actual channel count to use for this source. Three factors at
  // play here: the device capabilities, the constraints passed in by content,
  // and a pref that can force things (for testing)
  int32_t maxChannels = mDeviceInfo->MaxChannels();

  // First, check channelCount violation wrt constraints. This fails in case of
  // error.
  if (c.mChannelCount.mMin > maxChannels) {
    *aOutBadConstraint = "channelCount";
    return NS_ERROR_FAILURE;
  }
  // A pref can force the channel count to use. If the pref has a value of zero
  // or lower, it has no effect.
  if (aInPrefs.mChannels <= 0) {
    prefs.mChannels = maxChannels;
  }

  // Get the number of channels asked for by content, and clamp it between the
  // pref and the maximum number of channels that the device supports.
  prefs.mChannels =
    c.mChannelCount.Get(std::min(aInPrefs.mChannels, maxChannels));
  prefs.mChannels = std::max(1, std::min(prefs.mChannels, maxChannels));

  LOG(("Audio config: aec: %d, agc: %d, noise: %d, channels: %d",
       prefs.mAecOn ? prefs.mAec : -1,
       prefs.mAgcOn ? prefs.mAgc : -1,
       prefs.mNoiseOn ? prefs.mNoise : -1,
       prefs.mChannels));

  *aOutPrefs = prefs;

  return NS_OK;
}

nsresult
MediaEngineWebRTCMicrophoneSource::Reconfigure(
  const RefPtr<AllocationHandle>&,
  const dom::MediaTrackConstraints& aConstraints,
  const MediaEnginePrefs& aPrefs,
  const nsString& /* aDeviceId */,
  const char** aOutBadConstraint)
{
  AssertIsOnOwningThread();
  MOZ_ASSERT(mStream);

  LOG(("Mic source %p Reconfigure ", this));

  NormalizedConstraints constraints(aConstraints);
  MediaEnginePrefs outputPrefs;
  nsresult rv =
    EvaluateSettings(constraints, aPrefs, &outputPrefs, aOutBadConstraint);
  if (NS_FAILED(rv)) {
    if (aOutBadConstraint) {
      return NS_ERROR_INVALID_ARG;
    }

    nsAutoCString name;
    GetErrorName(rv, name);
    LOG(("Mic source %p Reconfigure() failed unexpectedly. rv=%s",
         this,
         name.Data()));
    Stop(nullptr);
    return NS_ERROR_UNEXPECTED;
  }

  ApplySettings(outputPrefs);

  return NS_OK;
}

void
MediaEngineWebRTCMicrophoneSource::Pull(
  const RefPtr<const AllocationHandle>&,
  const RefPtr<SourceMediaStream>& aStream,
  TrackID aTrackID,
  StreamTime aDesiredTime,
  const PrincipalHandle& aPrincipalHandle)
{
  // If pull is enabled, it means that the audio input is not open, and we
  // should fill it out with silence. This is the only method called on the
  // MSG thread.
  mInputProcessing->Pull(aStream, aTrackID, aDesiredTime, aPrincipalHandle);
}

void
MediaEngineWebRTCMicrophoneSource::UpdateAECSettingsIfNeeded(
  bool aEnable,
  webrtc::EcModes aMode)
{
  AssertIsOnOwningThread();

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> gripGraph = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom(
    [that, graph = std::move(gripGraph), aEnable, aMode]() mutable {
      class Message : public ControlMessage
      {
      public:
        Message(AudioInputProcessing* aInputProcessing,
                bool aEnable,
                webrtc::EcModes aMode)
          : ControlMessage(nullptr)
          , mInputProcessing(aInputProcessing)
          , mEnable(aEnable)
          , mMode(aMode)
        {
        }

        void Run() override
        {
          mInputProcessing->UpdateAECSettingsIfNeeded(mEnable, mMode);
        }

      protected:
        RefPtr<AudioInputProcessing> mInputProcessing;
        bool mEnable;
        webrtc::EcModes mMode;
      };

      if (graph) {
        graph->AppendMessage(
          MakeUnique<Message>(that->mInputProcessing, aEnable, aMode));
      }

      return NS_OK;
    }));
}

void
MediaEngineWebRTCMicrophoneSource::UpdateAGCSettingsIfNeeded(
  bool aEnable,
  webrtc::AgcModes aMode)
{
  AssertIsOnOwningThread();

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> gripGraph = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom(
    [that, graph = std::move(gripGraph), aEnable, aMode]() mutable {
      class Message : public ControlMessage
      {
      public:
        Message(AudioInputProcessing* aInputProcessing,
                bool aEnable,
                webrtc::AgcModes aMode)
          : ControlMessage(nullptr)
          , mInputProcessing(aInputProcessing)
          , mEnable(aEnable)
          , mMode(aMode)
        {
        }

        void Run() override
        {
          mInputProcessing->UpdateAGCSettingsIfNeeded(mEnable, mMode);
        }

      protected:
        RefPtr<AudioInputProcessing> mInputProcessing;
        bool mEnable;
        webrtc::AgcModes mMode;
      };

      if (graph) {
        graph->AppendMessage(
          MakeUnique<Message>(that->mInputProcessing, aEnable, aMode));
      }

      return NS_OK;
    }));
}

void
MediaEngineWebRTCMicrophoneSource::UpdateNSSettingsIfNeeded(
  bool aEnable,
  webrtc::NsModes aMode)
{
  AssertIsOnOwningThread();

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> gripGraph = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom(
    [that, graph = std::move(gripGraph), aEnable, aMode]() mutable {
      class Message : public ControlMessage
      {
      public:
        Message(AudioInputProcessing* aInputProcessing,
                bool aEnable,
                webrtc::NsModes aMode)
          : ControlMessage(nullptr)
          , mInputProcessing(aInputProcessing)
          , mEnable(aEnable)
          , mMode(aMode)
        {
        }

        void Run() override
        {
          mInputProcessing->UpdateNSSettingsIfNeeded(mEnable, mMode);
        }

      protected:
        RefPtr<AudioInputProcessing> mInputProcessing;
        bool mEnable;
        webrtc::NsModes mMode;
      };

      if (graph) {
        graph->AppendMessage(
          MakeUnique<Message>(that->mInputProcessing, aEnable, aMode));
      }

      return NS_OK;
    }));
}

void
MediaEngineWebRTCMicrophoneSource::UpdateAPMExtraOptions(bool aExtendedFilter,
                                                         bool aDelayAgnostic)
{
  AssertIsOnOwningThread();

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> gripGraph = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom([that,
                                                  graph = std::move(gripGraph),
                                                  aExtendedFilter,
                                                  aDelayAgnostic]() mutable {
    class Message : public ControlMessage
    {
    public:
      Message(AudioInputProcessing* aInputProcessing,
              bool aExtendedFilter,
              bool aDelayAgnostic)
        : ControlMessage(nullptr)
        , mInputProcessing(aInputProcessing)
        , mExtendedFilter(aExtendedFilter)
        , mDelayAgnostic(aDelayAgnostic)
      {
      }

      void Run() override
      {
        mInputProcessing->UpdateAPMExtraOptions(mExtendedFilter,
                                                mDelayAgnostic);
      }

    protected:
      RefPtr<AudioInputProcessing> mInputProcessing;
      bool mExtendedFilter;
      bool mDelayAgnostic;
    };

    if (graph) {
      graph->AppendMessage(MakeUnique<Message>(
        that->mInputProcessing, aExtendedFilter, aDelayAgnostic));
    }

    return NS_OK;
  }));
}

void
MediaEngineWebRTCMicrophoneSource::ApplySettings(const MediaEnginePrefs& aPrefs)
{
  AssertIsOnOwningThread();

  MOZ_ASSERT(
    mStream,
    "ApplySetting is to be called only after SetTrack has been called");

  if (mStream) {
    UpdateAGCSettingsIfNeeded(aPrefs.mAgcOn,
                              static_cast<AgcModes>(aPrefs.mAgc));
    UpdateNSSettingsIfNeeded(aPrefs.mNoiseOn,
                             static_cast<NsModes>(aPrefs.mNoise));
    UpdateAECSettingsIfNeeded(aPrefs.mAecOn, static_cast<EcModes>(aPrefs.mAec));

    UpdateAPMExtraOptions(mExtendedFilter, mDelayAgnostic);
  }

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> graphImpl = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom(
    [that, graph = std::move(graphImpl), prefs = aPrefs]() mutable {
      that->mSettings->mEchoCancellation.Value() = prefs.mAecOn;
      that->mSettings->mAutoGainControl.Value() = prefs.mAgcOn;
      that->mSettings->mNoiseSuppression.Value() = prefs.mNoiseOn;
      that->mSettings->mChannelCount.Value() = prefs.mChannels;

      class Message : public ControlMessage
      {
      public:
        Message(AudioInputProcessing* aInputProcessing,
                bool aPassThrough,
                uint32_t aRequestedInputChannelCount)
          : ControlMessage(nullptr)
          , mInputProcessing(aInputProcessing)
          , mPassThrough(aPassThrough)
          , mRequestedInputChannelCount(aRequestedInputChannelCount)
        {
        }

        void Run() override
        {
          mInputProcessing->SetPassThrough(mPassThrough);
          mInputProcessing->SetRequestedInputChannelCount(
            mRequestedInputChannelCount);
        }

      protected:
        RefPtr<AudioInputProcessing> mInputProcessing;
        bool mPassThrough;
        uint32_t mRequestedInputChannelCount;
      };

      bool passThrough = !(prefs.mAecOn || prefs.mAgcOn || prefs.mNoiseOn);
      if (graph) {
        graph->AppendMessage(MakeUnique<Message>(
          that->mInputProcessing, passThrough, prefs.mChannels));
      }

      return NS_OK;
    }));
}

nsresult
MediaEngineWebRTCMicrophoneSource::Allocate(
  const dom::MediaTrackConstraints& aConstraints,
  const MediaEnginePrefs& aPrefs,
  const nsString& aDeviceId,
  const ipc::PrincipalInfo& aPrincipalInfo,
  AllocationHandle** aOutHandle,
  const char** aOutBadConstraint)
{
  AssertIsOnOwningThread();
  MOZ_ASSERT(aOutHandle);

  *aOutHandle = nullptr;

  mState = kAllocated;

  NormalizedConstraints normalized(aConstraints);
  MediaEnginePrefs outputPrefs;
  nsresult rv =
    EvaluateSettings(normalized, aPrefs, &outputPrefs, aOutBadConstraint);
  if (NS_FAILED(rv)) {
    return rv;
  }

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  NS_DispatchToMainThread(
    media::NewRunnableFrom([that, prefs = outputPrefs]() mutable {
      that->mSettings->mEchoCancellation.Value() = prefs.mAecOn;
      that->mSettings->mAutoGainControl.Value() = prefs.mAgcOn;
      that->mSettings->mNoiseSuppression.Value() = prefs.mNoiseOn;
      that->mSettings->mChannelCount.Value() = prefs.mChannels;
      return NS_OK;
    }));

  return rv;
}

nsresult
MediaEngineWebRTCMicrophoneSource::Deallocate(
  const RefPtr<const AllocationHandle>&)
{
  AssertIsOnOwningThread();

  MOZ_ASSERT(mState == kStopped);

  class EndTrackMessage : public ControlMessage
  {
  public:
    EndTrackMessage(MediaStream* aStream,
                    AudioInputProcessing* aAudioInputProcessing,
                    TrackID aTrackID)
      : ControlMessage(aStream)
      , mInputProcessing(aAudioInputProcessing)
      , mTrackID(aTrackID)
    {
    }

    void Run() override
    {
      mInputProcessing->End();
      mStream->AsSourceStream()->EndTrack(mTrackID);
    }

  protected:
    RefPtr<AudioInputProcessing> mInputProcessing;
    TrackID mTrackID;
  };

  if (mStream && IsTrackIDExplicit(mTrackID)) {
    RefPtr<MediaStream> sourceStream = mStream;
    RefPtr<MediaStreamGraphImpl> graphImpl = mStream->GraphImpl();
    RefPtr<AudioInputProcessing> inputProcessing = mInputProcessing;
    NS_DispatchToMainThread(
      media::NewRunnableFrom([graph = std::move(graphImpl),
                              stream = std::move(sourceStream),
                              audioInputProcessing = std::move(inputProcessing),
                              trackID = mTrackID]() mutable {
        if (graph) {
          graph->AppendMessage(
            MakeUnique<EndTrackMessage>(stream, audioInputProcessing, trackID));
        }
        return NS_OK;
      }));
  }

  MOZ_ASSERT(mTrackID != TRACK_NONE, "Only deallocate once");

  // Reset all state. This is not strictly necessary, this instance will get
  // destroyed soon.
  mStream = nullptr;
  mTrackID = TRACK_NONE;
  mPrincipal = PRINCIPAL_HANDLE_NONE;

  // If empty, no callbacks to deliver data should be occuring
  MOZ_ASSERT(mState != kReleased, "Source not allocated");
  MOZ_ASSERT(mState != kStarted, "Source not stopped");

  mState = kReleased;
  LOG(
    ("Audio device %s deallocated", NS_ConvertUTF16toUTF8(mDeviceName).get()));

  return NS_OK;
}

nsresult
MediaEngineWebRTCMicrophoneSource::SetTrack(
  const RefPtr<const AllocationHandle>&,
  const RefPtr<SourceMediaStream>& aStream,
  TrackID aTrackID,
  const PrincipalHandle& aPrincipal)
{
  AssertIsOnOwningThread();
  MOZ_ASSERT(aStream);
  MOZ_ASSERT(IsTrackIDExplicit(aTrackID));

  if (mStream && mStream->Graph() != aStream->Graph()) {
    return NS_ERROR_NOT_AVAILABLE;
  }

  MOZ_ASSERT(!mStream);
  MOZ_ASSERT(mTrackID == TRACK_NONE);
  MOZ_ASSERT(mPrincipal == PRINCIPAL_HANDLE_NONE);
  mStream = aStream;
  mTrackID = aTrackID;
  mPrincipal = aPrincipal;

  AudioSegment* segment = new AudioSegment();

  aStream->AddAudioTrack(aTrackID,
                         aStream->GraphRate(),
                         0,
                         segment,
                         SourceMediaStream::ADDTRACK_QUEUED);

  LOG(("Stream %p registered for microphone capture", aStream.get()));
  return NS_OK;
}

class StartStopMessage : public ControlMessage
{
public:
  enum StartStop
  {
    Start,
    Stop
  };

  StartStopMessage(AudioInputProcessing* aInputProcessing, StartStop aAction)
    : ControlMessage(nullptr)
    , mInputProcessing(aInputProcessing)
    , mAction(aAction)
  {
  }

  void Run() override
  {
    if (mAction == StartStopMessage::Start) {
      mInputProcessing->Start();
    } else if (mAction == StartStopMessage::Stop) {
      mInputProcessing->Stop();
    } else {
      MOZ_CRASH("Invalid enum value");
    }
  }

protected:
  RefPtr<AudioInputProcessing> mInputProcessing;
  StartStop mAction;
};

nsresult
MediaEngineWebRTCMicrophoneSource::Start(const RefPtr<const AllocationHandle>&)
{
  AssertIsOnOwningThread();

  // This spans setting both the enabled state and mState.
  if (mState == kStarted) {
    return NS_OK;
  }

  MOZ_ASSERT(mState == kAllocated || mState == kStopped);

  CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
  if (mStream->GraphImpl()->InputDeviceID() &&
      mStream->GraphImpl()->InputDeviceID() != deviceID) {
    // For now, we only allow opening a single audio input device per document,
    // because we can only have one MSG per document.
    return NS_ERROR_FAILURE;
  }

  // On Linux with PulseAudio, we still only allow a certain number of audio
  // input stream in each content process, because of issues related to audio
  // remoting and PulseAudio.
#ifdef MOZ_PULSEAUDIO
  // When remoting, cubeb reports it's using the "remote" backend instead of the
  // backend on the other side of the IPC.
  const char* backend = cubeb_get_backend_id(CubebUtils::GetCubebContext());
  if (strstr(backend, "remote") &&
      sInputStreamsOpen == CubebUtils::GetMaxInputStreams()) {
    LOG(("%p Already capturing audio in this process, aborting", this));
    return NS_ERROR_FAILURE;
  }

  sInputStreamsOpen++;
#endif

  AssertIsOnOwningThread();

  mInputProcessing = new AudioInputProcessing(
    mDeviceMaxChannelCount, mStream, mTrackID, mPrincipal);

  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> gripGraph = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom(
    [that, graph = std::move(gripGraph), deviceID]() mutable {
      if (graph) {
        graph->AppendMessage(MakeUnique<StartStopMessage>(
          that->mInputProcessing, StartStopMessage::Start));
      }

      that->mStream->OpenAudioInput(deviceID, that->mInputProcessing);

      return NS_OK;
    }));

  MOZ_ASSERT(mState != kReleased);
  mState = kStarted;

  return NS_OK;
}

nsresult
MediaEngineWebRTCMicrophoneSource::Stop(const RefPtr<const AllocationHandle>&)
{
  AssertIsOnOwningThread();

  LOG(("Mic source %p Stop()", this));

  MOZ_ASSERT(mStream, "SetTrack must have been called before ::Stop");

  if (mState == kStopped) {
    // Already stopped - this is allowed
    return NS_OK;
  }

#ifdef MOZ_PULSEAUDIO
  MOZ_ASSERT(sInputStreamsOpen > 0);
  sInputStreamsOpen--;
#endif
  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
  RefPtr<MediaStreamGraphImpl> gripGraph = mStream->GraphImpl();
  NS_DispatchToMainThread(media::NewRunnableFrom(
    [that, graph = std::move(gripGraph), stream = mStream]() mutable {
      if (graph) {
        graph->AppendMessage(MakeUnique<StartStopMessage>(
          that->mInputProcessing, StartStopMessage::Stop));
      }

      CubebUtils::AudioDeviceID deviceID = that->mDeviceInfo->DeviceID();
      Maybe<CubebUtils::AudioDeviceID> id = Some(deviceID);
      stream->CloseAudioInput(id, that->mInputProcessing);

      return NS_OK;
    }));

  MOZ_ASSERT(mState == kStarted, "Should be started when stopping");
  mState = kStopped;

  return NS_OK;
}

void
MediaEngineWebRTCMicrophoneSource::GetSettings(
  dom::MediaTrackSettings& aOutSettings) const
{
  MOZ_ASSERT(NS_IsMainThread());
  aOutSettings = *mSettings;
}

AudioInputProcessing::AudioInputProcessing(
  uint32_t aMaxChannelCount,
  RefPtr<SourceMediaStream> aStream,
  TrackID aTrackID,
  const PrincipalHandle& aPrincipalHandle)
  : mStream(std::move(aStream))
  , mAudioProcessing(AudioProcessing::Create())
  , mRequestedInputChannelCount(aMaxChannelCount)
  , mSkipProcessing(false)
  , mInputDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100)
#ifdef DEBUG
  , mLastCallbackAppendTime(0)
#endif
  , mLiveFramesAppended(false)
  , mLiveSilenceAppended(false)
  , mTrackID(aTrackID)
  , mPrincipal(aPrincipalHandle)
  , mEnabled(false)
  , mEnded(false)
{
}

void
AudioInputProcessing::Disconnect(MediaStreamGraphImpl* aGraph)
{
  // This method is just for asserts.
  MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
}

void
MediaEngineWebRTCMicrophoneSource::Shutdown()
{
  AssertIsOnOwningThread();

  if (mState == kStarted) {
    Stop(nullptr);
    MOZ_ASSERT(mState == kStopped);
  }

  MOZ_ASSERT(mState == kAllocated || mState == kStopped);
  Deallocate(nullptr);
  MOZ_ASSERT(mState == kReleased);
}

bool
AudioInputProcessing::PassThrough(MediaStreamGraphImpl* aGraph) const
{
  MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
  return mSkipProcessing;
}

void
AudioInputProcessing::SetPassThrough(bool aPassThrough)
{
  mSkipProcessing = aPassThrough;
}

uint32_t
AudioInputProcessing::GetRequestedInputChannelCount(
  MediaStreamGraphImpl* aGraphImpl)
{
  return mRequestedInputChannelCount;
}

void
AudioInputProcessing::SetRequestedInputChannelCount(
  uint32_t aRequestedInputChannelCount)
{
  mRequestedInputChannelCount = aRequestedInputChannelCount;

  mStream->GraphImpl()->ReevaluateInputDevice();
}

// This does an early return in case of error.
#define HANDLE_APM_ERROR(fn)                                                   \
  do {                                                                         \
    int rv = fn;                                                               \
    if (rv != AudioProcessing::kNoError) {                                     \
      MOZ_ASSERT_UNREACHABLE("APM error in " #fn);                             \
      return;                                                                  \
    }                                                                          \
  } while (0);

void
AudioInputProcessing::UpdateAECSettingsIfNeeded(bool aEnable, EcModes aMode)
{
  using webrtc::EcModes;

  EchoCancellation::SuppressionLevel level;

  switch (aMode) {
    case EcModes::kEcUnchanged:
      level = mAudioProcessing->echo_cancellation()->suppression_level();
      break;
    case EcModes::kEcConference:
      level = EchoCancellation::kHighSuppression;
      break;
    case EcModes::kEcDefault:
      level = EchoCancellation::kModerateSuppression;
      break;
    case EcModes::kEcAec:
      level = EchoCancellation::kModerateSuppression;
      break;
    case EcModes::kEcAecm:
      // No suppression level to set for the mobile echo canceller
      break;
    default:
      MOZ_LOG(GetMediaManagerLog(), LogLevel::Error, ("Bad EcMode value"));
      MOZ_ASSERT_UNREACHABLE("Bad pref set in all.js or in about:config"
                             " for the echo cancelation mode.");
      // fall back to something sensible in release
      level = EchoCancellation::kModerateSuppression;
      break;
  }

  // AECm and AEC are mutually exclusive.
  if (aMode == EcModes::kEcAecm) {
    HANDLE_APM_ERROR(mAudioProcessing->echo_cancellation()->Enable(false));
    HANDLE_APM_ERROR(mAudioProcessing->echo_control_mobile()->Enable(aEnable));
  } else {
    HANDLE_APM_ERROR(mAudioProcessing->echo_control_mobile()->Enable(false));
    HANDLE_APM_ERROR(mAudioProcessing->echo_cancellation()->Enable(aEnable));
    HANDLE_APM_ERROR(
      mAudioProcessing->echo_cancellation()->set_suppression_level(level));
  }
}

void
AudioInputProcessing::UpdateAGCSettingsIfNeeded(bool aEnable, AgcModes aMode)
{
#if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
  if (aMode == kAgcAdaptiveAnalog) {
    MOZ_LOG(GetMediaManagerLog(),
            LogLevel::Error,
            ("Invalid AGC mode kAgcAdaptiveAnalog on mobile"));
    MOZ_ASSERT_UNREACHABLE("Bad pref set in all.js or in about:config"
                           " for the auto gain, on mobile.");
    aMode = kAgcDefault;
  }
#endif
  GainControl::Mode mode = kDefaultAgcMode;

  switch (aMode) {
    case AgcModes::kAgcDefault:
      mode = kDefaultAgcMode;
      break;
    case AgcModes::kAgcUnchanged:
      mode = mAudioProcessing->gain_control()->mode();
      break;
    case AgcModes::kAgcFixedDigital:
      mode = GainControl::Mode::kFixedDigital;
      break;
    case AgcModes::kAgcAdaptiveAnalog:
      mode = GainControl::Mode::kAdaptiveAnalog;
      break;
    case AgcModes::kAgcAdaptiveDigital:
      mode = GainControl::Mode::kAdaptiveDigital;
      break;
    default:
      MOZ_ASSERT_UNREACHABLE("Bad pref set in all.js or in about:config"
                             " for the auto gain.");
      // This is a good fallback, it works regardless of the platform.
      mode = GainControl::Mode::kAdaptiveDigital;
      break;
  }

  HANDLE_APM_ERROR(mAudioProcessing->gain_control()->set_mode(mode));
  HANDLE_APM_ERROR(mAudioProcessing->gain_control()->Enable(aEnable));
}

void
AudioInputProcessing::UpdateNSSettingsIfNeeded(bool aEnable, NsModes aMode)
{
  NoiseSuppression::Level nsLevel;

  switch (aMode) {
    case NsModes::kNsDefault:
      nsLevel = kDefaultNsMode;
      break;
    case NsModes::kNsUnchanged:
      nsLevel = mAudioProcessing->noise_suppression()->level();
      break;
    case NsModes::kNsConference:
      nsLevel = NoiseSuppression::kHigh;
      break;
    case NsModes::kNsLowSuppression:
      nsLevel = NoiseSuppression::kLow;
      break;
    case NsModes::kNsModerateSuppression:
      nsLevel = NoiseSuppression::kModerate;
      break;
    case NsModes::kNsHighSuppression:
      nsLevel = NoiseSuppression::kHigh;
      break;
    case NsModes::kNsVeryHighSuppression:
      nsLevel = NoiseSuppression::kVeryHigh;
      break;
    default:
      MOZ_ASSERT_UNREACHABLE("Bad pref set in all.js or in about:config"
                             " for the noise suppression.");
      // Pick something sensible as a faillback in release.
      nsLevel = NoiseSuppression::kModerate;
  }
  HANDLE_APM_ERROR(mAudioProcessing->noise_suppression()->set_level(nsLevel));
  HANDLE_APM_ERROR(mAudioProcessing->noise_suppression()->Enable(aEnable));
}

#undef HANDLE_APM_ERROR

void
AudioInputProcessing::UpdateAPMExtraOptions(bool aExtendedFilter,
                                            bool aDelayAgnostic)
{
  webrtc::Config config;
  config.Set<webrtc::ExtendedFilter>(
    new webrtc::ExtendedFilter(aExtendedFilter));
  config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(aDelayAgnostic));

  mAudioProcessing->SetExtraOptions(config);
}

void
AudioInputProcessing::Start()
{
  mEnabled = true;
}

void
AudioInputProcessing::Stop()
{
  mEnabled = false;
}

void
AudioInputProcessing::Pull(const RefPtr<SourceMediaStream>& aStream,
                           TrackID aTrackID,
                           StreamTime aDesiredTime,
                           const PrincipalHandle& aPrincipalHandle)
{
  TRACE_AUDIO_CALLBACK_COMMENT(
    "SourceMediaStream %p track %i", aStream.get(), aTrackID);
  StreamTime delta;

  if (mEnded) {
    return;
  }

  delta = aDesiredTime - aStream->GetEndOfAppendedData(aTrackID);

  if (delta < 0) {
    LOG_FRAMES(
      ("Not appending silence; %" PRId64 " frames already buffered", -delta));
    return;
  }

  if (!mLiveFramesAppended || !mLiveSilenceAppended) {
    // These are the iterations after starting or resuming audio capture.
    // Make sure there's at least one extra block buffered until audio
    // callbacks come in. We also allow appending silence one time after
    // audio callbacks have started, to cover the case where audio callbacks
    // start appending data immediately and there is no extra data buffered.
    delta += WEBAUDIO_BLOCK_SIZE;

    // If we're supposed to be packetizing but there's no packetizer yet,
    // there must not have been any live frames appended yet.
    // If there were live frames appended and we haven't appended the
    // right amount of silence, we'll have to append silence once more,
    // failing the other assert below.
    MOZ_ASSERT_IF(!PassThrough(aStream->GraphImpl()) && !mPacketizerInput,
                  !mLiveFramesAppended);

    if (!PassThrough(aStream->GraphImpl()) && mPacketizerInput) {
      // Processing is active and is processed in chunks of 10ms through the
      // input packetizer. We allow for 10ms of silence on the track to
      // accomodate the buffering worst-case.
      delta += mPacketizerInput->PacketSize();
    }
  }

  LOG_FRAMES(("Pulling %" PRId64 " frames of silence.", delta));

  // This assertion fails when we append silence here in the same iteration
  // as there were real audio samples already appended by the audio callback.
  // Note that this is exempted until live samples and a subsequent chunk of
  // silence have been appended to the track. This will cover cases like:
  // - After Start(), there is silence (maybe multiple times) appended before
  //   the first audio callback.
  // - After Start(), there is real data (maybe multiple times) appended
  //   before the first graph iteration.
  // And other combinations of order of audio sample sources.
  MOZ_ASSERT_IF(mEnabled && mLiveFramesAppended && mLiveSilenceAppended,
                aStream->GraphImpl()->IterationEnd() > mLastCallbackAppendTime);

  if (mLiveFramesAppended) {
    mLiveSilenceAppended = true;
  }

  AudioSegment audio;
  audio.AppendNullData(delta);
  aStream->AppendToTrack(aTrackID, &audio);
}

void
AudioInputProcessing::NotifyOutputData(MediaStreamGraphImpl* aGraph,
                                       AudioDataValue* aBuffer,
                                       size_t aFrames,
                                       TrackRate aRate,
                                       uint32_t aChannels)
{
  MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
  MOZ_ASSERT(mEnabled);

  if (!mPacketizerOutput || mPacketizerOutput->PacketSize() != aRate / 100u ||
      mPacketizerOutput->Channels() != aChannels) {
    // It's ok to drop the audio still in the packetizer here: if this changes,
    // we changed devices or something.
    mPacketizerOutput =
      new AudioPacketizer<AudioDataValue, float>(aRate / 100, aChannels);
  }

  mPacketizerOutput->Input(aBuffer, aFrames);

  while (mPacketizerOutput->PacketsAvailable()) {
    uint32_t samplesPerPacket =
      mPacketizerOutput->PacketSize() * mPacketizerOutput->Channels();
    if (mOutputBuffer.Length() < samplesPerPacket) {
      mOutputBuffer.SetLength(samplesPerPacket);
    }
    if (mDeinterleavedBuffer.Length() < samplesPerPacket) {
      mDeinterleavedBuffer.SetLength(samplesPerPacket);
    }
    float* packet = mOutputBuffer.Data();
    mPacketizerOutput->Output(packet);

    AutoTArray<float*, MAX_CHANNELS> deinterleavedPacketDataChannelPointers;
    float* interleavedFarend = nullptr;
    uint32_t channelCountFarend = 0;
    uint32_t framesPerPacketFarend = 0;

    // Downmix from aChannels to MAX_CHANNELS if needed. We always have floats
    // here, the packetized performed the conversion.
    if (aChannels > MAX_CHANNELS) {
      AudioConverter converter(
        AudioConfig(aChannels, 0, AudioConfig::FORMAT_FLT),
        AudioConfig(MAX_CHANNELS, 0, AudioConfig::FORMAT_FLT));
      framesPerPacketFarend = mPacketizerOutput->PacketSize();
      framesPerPacketFarend =
        converter.Process(mInputDownmixBuffer, packet, framesPerPacketFarend);
      interleavedFarend = mInputDownmixBuffer.Data();
      channelCountFarend = MAX_CHANNELS;
      deinterleavedPacketDataChannelPointers.SetLength(MAX_CHANNELS);
    } else {
      interleavedFarend = packet;
      channelCountFarend = aChannels;
      framesPerPacketFarend = mPacketizerOutput->PacketSize();
      deinterleavedPacketDataChannelPointers.SetLength(aChannels);
    }

    MOZ_ASSERT(interleavedFarend &&
               (channelCountFarend == 1 || channelCountFarend == 2) &&
               framesPerPacketFarend);

    if (mInputBuffer.Length() < framesPerPacketFarend * channelCountFarend) {
      mInputBuffer.SetLength(framesPerPacketFarend * channelCountFarend);
    }

    size_t offset = 0;
    for (size_t i = 0; i < deinterleavedPacketDataChannelPointers.Length();
         ++i) {
      deinterleavedPacketDataChannelPointers[i] = mInputBuffer.Data() + offset;
      offset += framesPerPacketFarend;
    }

    // Deinterleave, prepare a channel pointers array, with enough storage for
    // the frames.
    DeinterleaveAndConvertBuffer(
      interleavedFarend,
      framesPerPacketFarend,
      channelCountFarend,
      deinterleavedPacketDataChannelPointers.Elements());

    // Having the same config for input and output means we potentially save
    // some CPU.
    StreamConfig inputConfig(aRate, channelCountFarend, false);
    StreamConfig outputConfig = inputConfig;

    // Passing the same pointers here saves a copy inside this function.
    DebugOnly<int> err = mAudioProcessing->ProcessReverseStream(
      deinterleavedPacketDataChannelPointers.Elements(),
      inputConfig,
      outputConfig,
      deinterleavedPacketDataChannelPointers.Elements());

    MOZ_ASSERT(!err, "Could not process the reverse stream.");
  }
}

// Only called if we're not in passthrough mode
void
AudioInputProcessing::PacketizeAndProcess(MediaStreamGraphImpl* aGraph,
                                          const AudioDataValue* aBuffer,
                                          size_t aFrames,
                                          TrackRate aRate,
                                          uint32_t aChannels)
{
  MOZ_ASSERT(!PassThrough(aGraph),
             "This should be bypassed when in PassThrough mode.");
  MOZ_ASSERT(mEnabled);
  size_t offset = 0;

  if (!mPacketizerInput || mPacketizerInput->PacketSize() != aRate / 100u ||
      mPacketizerInput->Channels() != aChannels) {
    // It's ok to drop the audio still in the packetizer here.
    mPacketizerInput =
      new AudioPacketizer<AudioDataValue, float>(aRate / 100, aChannels);
  }

  // Packetize our input data into 10ms chunks, deinterleave into planar channel
  // buffers, process, and append to the right MediaStreamTrack.
  mPacketizerInput->Input(aBuffer, static_cast<uint32_t>(aFrames));

  while (mPacketizerInput->PacketsAvailable()) {
    uint32_t samplesPerPacket =
      mPacketizerInput->PacketSize() * mPacketizerInput->Channels();
    if (mInputBuffer.Length() < samplesPerPacket) {
      mInputBuffer.SetLength(samplesPerPacket);
    }
    if (mDeinterleavedBuffer.Length() < samplesPerPacket) {
      mDeinterleavedBuffer.SetLength(samplesPerPacket);
    }
    float* packet = mInputBuffer.Data();
    mPacketizerInput->Output(packet);

    // Deinterleave the input data
    // Prepare an array pointing to deinterleaved channels.
    AutoTArray<float*, 8> deinterleavedPacketizedInputDataChannelPointers;
    deinterleavedPacketizedInputDataChannelPointers.SetLength(aChannels);
    offset = 0;
    for (size_t i = 0;
         i < deinterleavedPacketizedInputDataChannelPointers.Length();
         ++i) {
      deinterleavedPacketizedInputDataChannelPointers[i] =
        mDeinterleavedBuffer.Data() + offset;
      offset += mPacketizerInput->PacketSize();
    }

    // Deinterleave to mInputBuffer, pointed to by inputBufferChannelPointers.
    Deinterleave(packet,
                 mPacketizerInput->PacketSize(),
                 aChannels,
                 deinterleavedPacketizedInputDataChannelPointers.Elements());

    StreamConfig inputConfig(
      aRate, aChannels, false /* we don't use typing detection*/);
    StreamConfig outputConfig = inputConfig;

    // Bug 1404965: Get the right delay here, it saves some work down the line.
    mAudioProcessing->set_stream_delay_ms(0);

    // Bug 1414837: find a way to not allocate here.
    RefPtr<SharedBuffer> buffer = SharedBuffer::Create(
      mPacketizerInput->PacketSize() * aChannels * sizeof(float));

    // Prepare channel pointers to the SharedBuffer created above.
    AutoTArray<float*, 8> processedOutputChannelPointers;
    AutoTArray<const float*, 8> processedOutputChannelPointersConst;
    processedOutputChannelPointers.SetLength(aChannels);
    processedOutputChannelPointersConst.SetLength(aChannels);

    offset = 0;
    for (size_t i = 0; i < processedOutputChannelPointers.Length(); ++i) {
      processedOutputChannelPointers[i] =
        static_cast<float*>(buffer->Data()) + offset;
      processedOutputChannelPointersConst[i] =
        static_cast<float*>(buffer->Data()) + offset;
      offset += mPacketizerInput->PacketSize();
    }

    mAudioProcessing->ProcessStream(
      deinterleavedPacketizedInputDataChannelPointers.Elements(),
      inputConfig,
      outputConfig,
      processedOutputChannelPointers.Elements());

    AudioSegment segment;
    if (!mStream->GraphImpl()) {
      // The DOMMediaStream that owns mStream has been cleaned up
      // and MediaStream::DestroyImpl() has run in the MSG. This is fine and
      // can happen before the MediaManager thread gets to stop capture for
      // this MediaStream.
      continue;
    }

    LOG_FRAMES(("Appending %" PRIu32 " frames of packetized audio",
                mPacketizerInput->PacketSize()));

#ifdef DEBUG
    mLastCallbackAppendTime = mStream->GraphImpl()->IterationEnd();
#endif
    mLiveFramesAppended = true;

    // We already have planar audio data of the right format. Insert into the
    // MSG.
    MOZ_ASSERT(processedOutputChannelPointers.Length() == aChannels);
    RefPtr<SharedBuffer> other = buffer;
    segment.AppendFrames(other.forget(),
                         processedOutputChannelPointersConst,
                         mPacketizerInput->PacketSize(),
                         mPrincipal);
    mStream->AppendToTrack(mTrackID, &segment);
  }
}

template<typename T>
void
AudioInputProcessing::InsertInGraph(const T* aBuffer,
                                    size_t aFrames,
                                    uint32_t aChannels)
{
  if (!mStream->GraphImpl()) {
    // The DOMMediaStream that owns mStream has been cleaned up
    // and MediaStream::DestroyImpl() has run in the MSG. This is fine and
    // can happen before the MediaManager thread gets to stop capture for
    // this MediaStream.
    return;
  }

#ifdef DEBUG
  mLastCallbackAppendTime = mStream->GraphImpl()->IterationEnd();
#endif
  mLiveFramesAppended = true;

  MOZ_ASSERT(aChannels >= 1 && aChannels <= 8, "Support up to 8 channels");

  AudioSegment segment;
  RefPtr<SharedBuffer> buffer =
    SharedBuffer::Create(aFrames * aChannels * sizeof(T));
  AutoTArray<const T*, 8> channels;
  if (aChannels == 1) {
    PodCopy(static_cast<T*>(buffer->Data()), aBuffer, aFrames);
    channels.AppendElement(static_cast<T*>(buffer->Data()));
  } else {
    channels.SetLength(aChannels);
    AutoTArray<T*, 8> write_channels;
    write_channels.SetLength(aChannels);
    T* samples = static_cast<T*>(buffer->Data());

    size_t offset = 0;
    for (uint32_t i = 0; i < aChannels; ++i) {
      channels[i] = write_channels[i] = samples + offset;
      offset += aFrames;
    }

    DeinterleaveAndConvertBuffer(
      aBuffer, aFrames, aChannels, write_channels.Elements());
  }

  LOG_FRAMES(("Appending %zu frames of raw audio", aFrames));

  MOZ_ASSERT(aChannels == channels.Length());
  segment.AppendFrames(buffer.forget(), channels, aFrames, mPrincipal);

  mStream->AppendToTrack(mTrackID, &segment);
}

// Called back on GraphDriver thread!
// Note this can be called back after ::Shutdown()
void
AudioInputProcessing::NotifyInputData(MediaStreamGraphImpl* aGraph,
                                      const AudioDataValue* aBuffer,
                                      size_t aFrames,
                                      TrackRate aRate,
                                      uint32_t aChannels)
{
  MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
  TRACE_AUDIO_CALLBACK();

  MOZ_ASSERT(mEnabled);

  // If some processing is necessary, packetize and insert in the WebRTC.org
  // code. Otherwise, directly insert the mic data in the MSG, bypassing all
  // processing.
  if (PassThrough(aGraph)) {
    InsertInGraph<AudioDataValue>(aBuffer, aFrames, aChannels);
  } else {
    PacketizeAndProcess(aGraph, aBuffer, aFrames, aRate, aChannels);
  }
}

#define ResetProcessingIfNeeded(_processing)                                   \
  do {                                                                         \
    bool enabled = mAudioProcessing->_processing()->is_enabled();              \
                                                                               \
    if (enabled) {                                                             \
      int rv = mAudioProcessing->_processing()->Enable(!enabled);              \
      if (rv) {                                                                \
        NS_WARNING("Could not reset the status of the " #_processing           \
                   " on device change.");                                      \
        return;                                                                \
      }                                                                        \
      rv = mAudioProcessing->_processing()->Enable(enabled);                   \
      if (rv) {                                                                \
        NS_WARNING("Could not reset the status of the " #_processing           \
                   " on device change.");                                      \
        return;                                                                \
      }                                                                        \
    }                                                                          \
  } while (0)

void
AudioInputProcessing::DeviceChanged(MediaStreamGraphImpl* aGraph)
{
  MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
  // Reset some processing
  ResetProcessingIfNeeded(gain_control);
  ResetProcessingIfNeeded(echo_cancellation);
  ResetProcessingIfNeeded(noise_suppression);
}

void
AudioInputProcessing::End()
{
  mEnded = true;
}

nsString
MediaEngineWebRTCAudioCaptureSource::GetName() const
{
  return NS_LITERAL_STRING(u"AudioCapture");
}

nsCString
MediaEngineWebRTCAudioCaptureSource::GetUUID() const
{
  nsID uuid;
  char uuidBuffer[NSID_LENGTH];
  nsCString asciiString;
  ErrorResult rv;

  rv = nsContentUtils::GenerateUUIDInPlace(uuid);
  if (rv.Failed()) {
    return NS_LITERAL_CSTRING("");
  }

  uuid.ToProvidedString(uuidBuffer);
  asciiString.AssignASCII(uuidBuffer);

  // Remove {} and the null terminator
  return nsCString(Substring(asciiString, 1, NSID_LENGTH - 3));
}

nsresult
MediaEngineWebRTCAudioCaptureSource::SetTrack(
  const RefPtr<const AllocationHandle>&,
  const RefPtr<SourceMediaStream>& aStream,
  TrackID aTrackID,
  const PrincipalHandle& aPrincipalHandle)
{
  AssertIsOnOwningThread();
  // Nothing to do here. aStream is a placeholder dummy and not exposed.
  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioCaptureSource::Start(
  const RefPtr<const AllocationHandle>&)
{
  AssertIsOnOwningThread();
  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioCaptureSource::Stop(const RefPtr<const AllocationHandle>&)
{
  AssertIsOnOwningThread();
  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioCaptureSource::Reconfigure(
  const RefPtr<AllocationHandle>&,
  const dom::MediaTrackConstraints& aConstraints,
  const MediaEnginePrefs& aPrefs,
  const nsString& aDeviceId,
  const char** aOutBadConstraint)
{
  return NS_OK;
}

uint32_t
MediaEngineWebRTCAudioCaptureSource::GetBestFitnessDistance(
  const nsTArray<const NormalizedConstraintSet*>& aConstraintSets,
  const nsString& aDeviceId) const
{
  // There is only one way of capturing audio for now, and it's always adequate.
  return 0;
}

}