/
MediaEngineWebRTCAudio.cpp
1329 lines (1133 loc) · 49.1 KB
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MediaEngineWebRTCAudio.cpp
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/* -*- 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 "MediaTrackGraphImpl.h"
#include "MediaTrackConstraints.h"
#include "mozilla/Assertions.h"
#include "mozilla/ErrorNames.h"
#include "nsIDUtils.h"
#include "transport/runnable_utils.h"
#include "Tracing.h"
#include "mozilla/Sprintf.h"
#include "mozilla/Logging.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/include/audio_processing.h"
using namespace webrtc;
// These are restrictions from the webrtc.org code
#define MAX_CHANNELS 2
#define MONO 1
#define MAX_SAMPLING_FREQ 48000 // Hz - multiple of 100
namespace mozilla {
using dom::MediaSourceEnum;
extern LazyLogModule gMediaManagerLog;
#define LOG(...) MOZ_LOG(gMediaManagerLog, LogLevel::Debug, (__VA_ARGS__))
#define LOG_FRAME(...) \
MOZ_LOG(gMediaManagerLog, LogLevel::Verbose, (__VA_ARGS__))
#define LOG_ERROR(...) MOZ_LOG(gMediaManagerLog, LogLevel::Error, (__VA_ARGS__))
/**
* WebRTC Microphone MediaEngineSource.
*/
MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
const MediaDevice* aMediaDevice)
: mPrincipal(PRINCIPAL_HANDLE_NONE),
mDeviceInfo(aMediaDevice->mAudioDeviceInfo),
mDeviceMaxChannelCount(mDeviceInfo->MaxChannels()),
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)) {
MOZ_ASSERT(aMediaDevice->mMediaSource == MediaSourceEnum::Microphone);
#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;
}
nsresult MediaEngineWebRTCMicrophoneSource::EvaluateSettings(
const NormalizedConstraints& aConstraintsUpdate,
const MediaEnginePrefs& aInPrefs, MediaEnginePrefs* aOutPrefs,
const char** aOutBadConstraint) {
AssertIsOnOwningThread();
FlattenedConstraints c(aConstraintsUpdate);
MediaEnginePrefs prefs = aInPrefs;
prefs.mAecOn = c.mEchoCancellation.Get(aInPrefs.mAecOn);
prefs.mAgcOn = c.mAutoGainControl.Get(aInPrefs.mAgcOn && prefs.mAecOn);
prefs.mNoiseOn = c.mNoiseSuppression.Get(aInPrefs.mNoiseOn && prefs.mAecOn);
// 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 = static_cast<int32_t>(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(prefs.mChannels, maxChannels));
prefs.mChannels = std::max(1, std::min(prefs.mChannels, maxChannels));
LOG("Audio config: agc: %d, noise: %d, channels: %d",
prefs.mAgcOn ? prefs.mAgc : -1, prefs.mNoiseOn ? prefs.mNoise : -1,
prefs.mChannels);
*aOutPrefs = prefs;
return NS_OK;
}
nsresult MediaEngineWebRTCMicrophoneSource::Reconfigure(
const dom::MediaTrackConstraints& aConstraints,
const MediaEnginePrefs& aPrefs, const char** aOutBadConstraint) {
AssertIsOnOwningThread();
MOZ_ASSERT(mTrack);
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();
return NS_ERROR_UNEXPECTED;
}
ApplySettings(outputPrefs);
mCurrentPrefs = outputPrefs;
return NS_OK;
}
void MediaEngineWebRTCMicrophoneSource::ApplySettings(
const MediaEnginePrefs& aPrefs) {
AssertIsOnOwningThread();
TRACE("ApplySettings");
MOZ_ASSERT(
mTrack,
"ApplySetting is to be called only after SetTrack has been called");
mAudioProcessingConfig.pipeline.multi_channel_render = true;
mAudioProcessingConfig.pipeline.multi_channel_capture = true;
mAudioProcessingConfig.echo_canceller.enabled = aPrefs.mAecOn;
mAudioProcessingConfig.echo_canceller.mobile_mode = aPrefs.mUseAecMobile;
if ((mAudioProcessingConfig.gain_controller1.enabled =
aPrefs.mAgcOn && !aPrefs.mAgc2Forced)) {
auto mode = static_cast<AudioProcessing::Config::GainController1::Mode>(
aPrefs.mAgc);
if (mode != AudioProcessing::Config::GainController1::kAdaptiveAnalog &&
mode != AudioProcessing::Config::GainController1::kAdaptiveDigital &&
mode != AudioProcessing::Config::GainController1::kFixedDigital) {
LOG_ERROR("AudioInputProcessing %p Attempt to set invalid AGC mode %d",
mInputProcessing.get(), static_cast<int>(mode));
mode = AudioProcessing::Config::GainController1::kAdaptiveDigital;
}
#if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
if (mode == AudioProcessing::Config::GainController1::kAdaptiveAnalog) {
LOG_ERROR(
"AudioInputProcessing %p Invalid AGC mode kAdaptiveAnalog on "
"mobile",
mInputProcessing.get());
MOZ_ASSERT_UNREACHABLE(
"Bad pref set in all.js or in about:config"
" for the auto gain, on mobile.");
mode = AudioProcessing::Config::GainController1::kFixedDigital;
}
#endif
mAudioProcessingConfig.gain_controller1.mode = mode;
}
mAudioProcessingConfig.gain_controller2.enabled =
mAudioProcessingConfig.gain_controller2.adaptive_digital.enabled =
aPrefs.mAgcOn && aPrefs.mAgc2Forced;
if ((mAudioProcessingConfig.noise_suppression.enabled = aPrefs.mNoiseOn)) {
auto level = static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
aPrefs.mNoise);
if (level != AudioProcessing::Config::NoiseSuppression::kLow &&
level != AudioProcessing::Config::NoiseSuppression::kModerate &&
level != AudioProcessing::Config::NoiseSuppression::kHigh &&
level != AudioProcessing::Config::NoiseSuppression::kVeryHigh) {
LOG_ERROR(
"AudioInputProcessing %p Attempt to set invalid noise suppression "
"level %d",
mInputProcessing.get(), static_cast<int>(level));
level = AudioProcessing::Config::NoiseSuppression::kModerate;
}
mAudioProcessingConfig.noise_suppression.level = level;
}
mAudioProcessingConfig.transient_suppression.enabled = aPrefs.mTransientOn;
mAudioProcessingConfig.high_pass_filter.enabled = aPrefs.mHPFOn;
mAudioProcessingConfig.residual_echo_detector.enabled =
aPrefs.mResidualEchoOn;
RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
NS_DispatchToMainThread(NS_NewRunnableFunction(
__func__, [this, that, deviceID, track = mTrack, prefs = aPrefs,
audioProcessingConfig = mAudioProcessingConfig] {
mSettings->mEchoCancellation.Value() = prefs.mAecOn;
mSettings->mAutoGainControl.Value() = prefs.mAgcOn;
mSettings->mNoiseSuppression.Value() = prefs.mNoiseOn;
mSettings->mChannelCount.Value() = prefs.mChannels;
class Message : public ControlMessage {
CubebUtils::AudioDeviceID mDeviceID;
const RefPtr<AudioInputProcessing> mInputProcessing;
const AudioProcessing::Config mAudioProcessingConfig;
const bool mPassThrough;
const uint32_t mRequestedInputChannelCount;
public:
Message(MediaTrack* aTrack, CubebUtils::AudioDeviceID aDeviceID,
AudioInputProcessing* aInputProcessing,
const AudioProcessing::Config& aAudioProcessingConfig,
bool aPassThrough, uint32_t aRequestedInputChannelCount)
: ControlMessage(aTrack),
mDeviceID(aDeviceID),
mInputProcessing(aInputProcessing),
mAudioProcessingConfig(aAudioProcessingConfig),
mPassThrough(aPassThrough),
mRequestedInputChannelCount(aRequestedInputChannelCount) {}
void Run() override {
mInputProcessing->ApplyConfig(mTrack->GraphImpl(),
mAudioProcessingConfig);
{
TRACE("SetRequestedInputChannelCount");
mInputProcessing->SetRequestedInputChannelCount(
mTrack->GraphImpl(), mDeviceID, mRequestedInputChannelCount);
}
{
TRACE("SetPassThrough")
mInputProcessing->SetPassThrough(mTrack->GraphImpl(),
mPassThrough);
}
}
};
// The high-pass filter is not taken into account when activating the
// pass through, since it's not controllable from content.
bool passThrough = !(prefs.mAecOn || prefs.mAgcOn || prefs.mNoiseOn);
if (track->IsDestroyed()) {
return;
}
track->GraphImpl()->AppendMessage(MakeUnique<Message>(
track, deviceID, mInputProcessing, audioProcessingConfig,
passThrough, prefs.mChannels));
}));
}
nsresult MediaEngineWebRTCMicrophoneSource::Allocate(
const dom::MediaTrackConstraints& aConstraints,
const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
const char** aOutBadConstraint) {
AssertIsOnOwningThread();
mState = kAllocated;
NormalizedConstraints normalized(aConstraints);
MediaEnginePrefs outputPrefs;
nsresult rv =
EvaluateSettings(normalized, aPrefs, &outputPrefs, aOutBadConstraint);
if (NS_FAILED(rv)) {
return rv;
}
NS_DispatchToMainThread(NS_NewRunnableFunction(
__func__, [settings = mSettings, prefs = outputPrefs] {
settings->mEchoCancellation.Value() = prefs.mAecOn;
settings->mAutoGainControl.Value() = prefs.mAgcOn;
settings->mNoiseSuppression.Value() = prefs.mNoiseOn;
settings->mChannelCount.Value() = prefs.mChannels;
}));
mCurrentPrefs = outputPrefs;
return rv;
}
nsresult MediaEngineWebRTCMicrophoneSource::Deallocate() {
AssertIsOnOwningThread();
MOZ_ASSERT(mState == kStopped || mState == kAllocated);
class EndTrackMessage : public ControlMessage {
const RefPtr<AudioInputProcessing> mInputProcessing;
public:
explicit EndTrackMessage(AudioInputProcessing* aAudioInputProcessing)
: ControlMessage(nullptr), mInputProcessing(aAudioInputProcessing) {}
void Run() override {
TRACE("mInputProcessing::End");
mInputProcessing->End();
}
};
if (mTrack) {
NS_DispatchToMainThread(NS_NewRunnableFunction(
__func__,
[track = std::move(mTrack), inputProcessing = mInputProcessing] {
if (track->IsDestroyed()) {
// This track has already been destroyed on main thread by its
// DOMMediaStream. No cleanup left to do.
return;
}
track->GraphImpl()->AppendMessage(
MakeUnique<EndTrackMessage>(inputProcessing));
}));
}
// Reset all state. This is not strictly necessary, this instance will get
// destroyed soon.
mTrack = nullptr;
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("Mic source %p Audio device %s deallocated", this,
NS_ConvertUTF16toUTF8(mDeviceInfo->Name()).get());
return NS_OK;
}
void MediaEngineWebRTCMicrophoneSource::SetTrack(
const RefPtr<MediaTrack>& aTrack, const PrincipalHandle& aPrincipal) {
AssertIsOnOwningThread();
MOZ_ASSERT(aTrack);
MOZ_ASSERT(aTrack->AsAudioProcessingTrack());
MOZ_ASSERT(!mTrack);
MOZ_ASSERT(mPrincipal == PRINCIPAL_HANDLE_NONE);
mTrack = aTrack->AsAudioProcessingTrack();
mPrincipal = aPrincipal;
mInputProcessing =
MakeAndAddRef<AudioInputProcessing>(mDeviceMaxChannelCount);
NS_DispatchToMainThread(NS_NewRunnableFunction(
__func__, [track = mTrack, processing = mInputProcessing]() mutable {
track->SetInputProcessing(std::move(processing));
track->Resume(); // Suspended by MediaManager
}));
LOG("Mic source %p Track %p registered for microphone capture", this,
aTrack.get());
}
class StartStopMessage : public ControlMessage {
public:
enum StartStop { Start, Stop };
StartStopMessage(MediaTrack* aTrack, AudioInputProcessing* aInputProcessing,
StartStop aAction)
: ControlMessage(aTrack),
mInputProcessing(aInputProcessing),
mAction(aAction) {}
void Run() override {
if (mAction == StartStopMessage::Start) {
TRACE("InputProcessing::Start")
mInputProcessing->Start(mTrack->GraphImpl());
} else if (mAction == StartStopMessage::Stop) {
TRACE("InputProcessing::Stop")
mInputProcessing->Stop(mTrack->GraphImpl());
} else {
MOZ_CRASH("Invalid enum value");
}
}
protected:
const RefPtr<AudioInputProcessing> mInputProcessing;
const StartStop mAction;
};
nsresult MediaEngineWebRTCMicrophoneSource::Start() {
AssertIsOnOwningThread();
// This spans setting both the enabled state and mState.
if (mState == kStarted) {
return NS_OK;
}
MOZ_ASSERT(mState == kAllocated || mState == kStopped);
ApplySettings(mCurrentPrefs);
CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
NS_DispatchToMainThread(NS_NewRunnableFunction(
__func__, [inputProcessing = mInputProcessing, deviceID, track = mTrack,
principal = mPrincipal] {
if (track->IsDestroyed()) {
return;
}
track->GraphImpl()->AppendMessage(MakeUnique<StartStopMessage>(
track, inputProcessing, StartStopMessage::Start));
track->ConnectDeviceInput(deviceID, inputProcessing.get(), principal);
}));
MOZ_ASSERT(mState != kReleased);
mState = kStarted;
return NS_OK;
}
nsresult MediaEngineWebRTCMicrophoneSource::Stop() {
AssertIsOnOwningThread();
LOG("Mic source %p Stop()", this);
MOZ_ASSERT(mTrack, "SetTrack must have been called before ::Stop");
if (mState == kStopped) {
// Already stopped - this is allowed
return NS_OK;
}
NS_DispatchToMainThread(NS_NewRunnableFunction(
__func__, [inputProcessing = mInputProcessing, deviceInfo = mDeviceInfo,
track = mTrack] {
if (track->IsDestroyed()) {
return;
}
MOZ_ASSERT(track->DeviceId().value() == deviceInfo->DeviceID());
track->DisconnectDeviceInput();
track->GraphImpl()->AppendMessage(MakeUnique<StartStopMessage>(
track, inputProcessing, StartStopMessage::Stop));
}));
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)
: mAudioProcessing(AudioProcessingBuilder().Create()),
mRequestedInputChannelCount(aMaxChannelCount),
mSkipProcessing(false),
mInputDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100),
mEnabled(false),
mEnded(false),
mPacketCount(0) {}
void AudioInputProcessing::Disconnect(MediaTrackGraphImpl* aGraph) {
// This method is just for asserts.
MOZ_ASSERT(aGraph->OnGraphThread());
}
bool AudioInputProcessing::PassThrough(MediaTrackGraphImpl* aGraph) const {
MOZ_ASSERT(aGraph->OnGraphThread());
return mSkipProcessing;
}
void AudioInputProcessing::SetPassThrough(MediaTrackGraphImpl* aGraph,
bool aPassThrough) {
MOZ_ASSERT(aGraph->OnGraphThread());
if (aPassThrough == mSkipProcessing) {
return;
}
mSkipProcessing = aPassThrough;
if (!mEnabled) {
MOZ_ASSERT(!mPacketizerInput);
return;
}
if (aPassThrough) {
// Turn on pass-through
ResetAudioProcessing(aGraph);
} else {
// Turn off pass-through
MOZ_ASSERT(!mPacketizerInput);
EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
}
}
uint32_t AudioInputProcessing::GetRequestedInputChannelCount() {
return mRequestedInputChannelCount;
}
void AudioInputProcessing::SetRequestedInputChannelCount(
MediaTrackGraphImpl* aGraph, CubebUtils::AudioDeviceID aDeviceId,
uint32_t aRequestedInputChannelCount) {
mRequestedInputChannelCount = aRequestedInputChannelCount;
aGraph->ReevaluateInputDevice(aDeviceId);
}
void AudioInputProcessing::Start(MediaTrackGraphImpl* aGraph) {
MOZ_ASSERT(aGraph->OnGraphThread());
if (mEnabled) {
return;
}
mEnabled = true;
if (mSkipProcessing) {
return;
}
MOZ_ASSERT(!mPacketizerInput);
EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
}
void AudioInputProcessing::Stop(MediaTrackGraphImpl* aGraph) {
MOZ_ASSERT(aGraph->OnGraphThread());
if (!mEnabled) {
return;
}
mEnabled = false;
if (mSkipProcessing) {
return;
}
// Packetizer is active and we were just stopped. Stop the packetizer and
// processing.
ResetAudioProcessing(aGraph);
}
// The following is how how Process() works in pass-through and non-pass-through
// mode. In both mode, Process() outputs the same amount of the frames as its
// input data.
//
// I. In non-pass-through mode:
//
// We will use webrtc::AudioProcessing to process the input audio data in this
// mode. The data input in webrtc::AudioProcessing needs to be a 10ms chunk,
// while the input data passed to Process() is not necessary to have times of
// 10ms-chunk length. To divide the input data into 10ms chunks,
// mPacketizerInput is introduced.
//
// We will add one 10ms-chunk silence into the internal buffer before Process()
// works. Those extra frames is called pre-buffering. It aims to avoid glitches
// we may have when producing data in mPacketizerInput. Without pre-buffering,
// when the input data length is not 10ms-times, we could end up having no
// enough output needs since mPacketizerInput would keep some input data, which
// is the remainder of the 10ms-chunk length. To force processing those data
// left in mPacketizerInput, we would need to add some extra frames to make
// mPacketizerInput produce a 10ms-chunk. For example, if the sample rate is
// 44100 Hz, then the packet-size is 441 frames. When we only have 384 input
// frames, we would need to put additional 57 frames to mPacketizerInput to
// produce a packet. However, those extra 57 frames result in a glitch sound.
//
// By adding one 10ms-chunk silence in advance to the internal buffer, we won't
// need to add extra frames between the input data no matter what data length it
// is. The only drawback is the input data won't be processed and send to output
// immediately. Process() will consume pre-buffering data for its output first.
// The below describes how it works:
//
//
// Process()
// +-----------------------------+
// input D(N) | +--------+ +--------+ | output D(N)
// --------------|-->| P(N) |-->| S(N) |---|-------------->
// | +--------+ +--------+ |
// | packetizer mSegment |
// +-----------------------------+
// <------ internal buffer ------>
//
//
// D(N): number of frames from the input and the output needs in the N round
// Z: number of frames of a 10ms chunk(packet) in mPacketizerInput, Z >= 1
// (if Z = 1, packetizer has no effect)
// P(N): number of frames left in mPacketizerInput after the N round. Once the
// frames in packetizer >= Z, packetizer will produce a packet to
// mSegment, so P(N) = (P(N-1) + D(N)) % Z, 0 <= P(N) <= Z-1
// S(N): number of frames left in mSegment after the N round. The input D(N)
// frames will be passed to mPacketizerInput first, and then
// mPacketizerInput may append some packets to mSegment, so
// S(N) = S(N-1) + Z * floor((P(N-1) + D(N)) / Z) - D(N)
//
// At the first, we set P(0) = 0, S(0) = X, where X >= Z-1. X is the
// pre-buffering put in the internal buffer. With this settings, P(K) + S(K) = X
// always holds.
//
// Intuitively, this seems true: We put X frames in the internal buffer at
// first. If the data won't be blocked in packetizer, after the Process(), the
// internal buffer should still hold X frames since the number of frames coming
// from input is the same as the output needs. The key of having enough data for
// output needs, while the input data is piled up in packetizer, is by putting
// at least Z-1 frames as pre-buffering, since the maximum number of frames
// stuck in the packetizer before it can emit a packet is packet-size - 1.
// Otherwise, we don't have enough data for output if the new input data plus
// the data left in packetizer produces a smaller-than-10ms chunk, which will be
// left in packetizer. Thus we must have some pre-buffering frames in the
// mSegment to make up the length of the left chunk we need for output. This can
// also be told by by induction:
// (1) This holds when K = 0
// (2) Assume this holds when K = N: so P(N) + S(N) = X
// => P(N) + S(N) = X >= Z-1 => S(N) >= Z-1-P(N)
// (3) When K = N+1, we have D(N+1) input frames comes
// a. if P(N) + D(N+1) < Z, then packetizer has no enough data for one
// packet. No data produced by packertizer, so the mSegment now has
// S(N) >= Z-1-P(N) frames. Output needs D(N+1) < Z-P(N) frames. So it
// needs at most Z-P(N)-1 frames, and mSegment has enough frames for
// output, Then, P(N+1) = P(N) + D(N+1) and S(N+1) = S(N) - D(N+1)
// => P(N+1) + S(N+1) = P(N) + S(N) = X
// b. if P(N) + D(N+1) = Z, then packetizer will produce one packet for
// mSegment, so mSegment now has S(N) + Z frames. Output needs D(N+1)
// = Z-P(N) frames. S(N) has at least Z-1-P(N)+Z >= Z-P(N) frames, since
// Z >= 1. So mSegment has enough frames for output. Then, P(N+1) = 0 and
// S(N+1) = S(N) + Z - D(N+1) = S(N) + P(N)
// => P(N+1) + S(N+1) = P(N) + S(N) = X
// c. if P(N) + D(N+1) > Z, and let P(N) + D(N+1) = q * Z + r, where q >= 1
// and 0 <= r <= Z-1, then packetizer will produce can produce q packets
// for mSegment. Output needs D(N+1) = q * Z - P(N) + r frames and
// mSegment has S(N) + q * z >= q * z - P(N) + Z-1 >= q*z -P(N) + r,
// since r <= Z-1. So mSegment has enough frames for output. Then,
// P(N+1) = r and S(N+1) = S(N) + q * Z - D(N+1)
// => P(N+1) + S(N+1) = S(N) + (q * Z + r - D(N+1)) = S(N) + P(N) = X
// => P(K) + S(K) = X always holds
//
// Since P(K) + S(K) = X and P(K) is in [0, Z-1], the S(K) is in [X-Z+1, X]
// range. In our implementation, X is set to Z so S(K) is in [1, Z].
// By the above workflow, we always have enough data for output and no extra
// frames put into packetizer. It means we don't have any glitch!
//
// II. In pass-through mode:
//
// Process()
// +--------+
// input D(N) | | output D(N)
// -------------->-------->--------------->
// | |
// +--------+
//
// The D(N) frames of data are just forwarded from input to output without any
// processing
void AudioInputProcessing::Process(MediaTrackGraphImpl* aGraph, GraphTime aFrom,
GraphTime aTo, AudioSegment* aInput,
AudioSegment* aOutput) {
MOZ_ASSERT(aGraph->OnGraphThread());
MOZ_ASSERT(aFrom <= aTo);
MOZ_ASSERT(!mEnded);
TrackTime need = aTo - aFrom;
if (need == 0) {
return;
}
if (!mEnabled) {
LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p Filling %" PRId64
" frames of silence to output (disabled)",
aGraph, aGraph->CurrentDriver(), this, need);
aOutput->AppendNullData(need);
return;
}
MOZ_ASSERT(aInput->GetDuration() == need,
"Wrong data length from input port source");
if (PassThrough(aGraph)) {
LOG_FRAME(
"(Graph %p, Driver %p) AudioInputProcessing %p Forwarding %" PRId64
" frames of input data to output directly (PassThrough)",
aGraph, aGraph->CurrentDriver(), this, aInput->GetDuration());
aOutput->AppendSegment(aInput);
return;
}
// SetPassThrough(false) must be called before reaching here.
MOZ_ASSERT(mPacketizerInput);
// If mRequestedInputChannelCount is updated, create a new packetizer. No
// need to change the pre-buffering since the rate is always the same. The
// frames left in the packetizer would be replaced by null data and then
// transferred to mSegment.
EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
// Preconditions of the audio-processing logic.
MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
mPacketizerInput->FramesAvailable() ==
mPacketizerInput->mPacketSize);
// We pre-buffer mPacketSize frames, but the maximum number of frames stuck in
// the packetizer before it can emit a packet is mPacketSize-1. Thus that
// remaining 1 frame will always be present in mSegment.
MOZ_ASSERT(mSegment.GetDuration() >= 1);
MOZ_ASSERT(mSegment.GetDuration() <= mPacketizerInput->mPacketSize);
PacketizeAndProcess(aGraph, *aInput);
LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p Buffer has %" PRId64
" frames of data now, after packetizing and processing",
aGraph, aGraph->CurrentDriver(), this, mSegment.GetDuration());
// By setting pre-buffering to the number of frames of one packet, and
// because the maximum number of frames stuck in the packetizer before
// it can emit a packet is the mPacketSize-1, we always have at least
// one more frame than output needs.
MOZ_ASSERT(mSegment.GetDuration() > need);
aOutput->AppendSlice(mSegment, 0, need);
mSegment.RemoveLeading(need);
LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p moving %" PRId64
" frames of data to output, leaving %" PRId64 " frames in buffer",
aGraph, aGraph->CurrentDriver(), this, need,
mSegment.GetDuration());
// Postconditions of the audio-processing logic.
MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
mPacketizerInput->FramesAvailable() ==
mPacketizerInput->mPacketSize);
MOZ_ASSERT(mSegment.GetDuration() >= 1);
MOZ_ASSERT(mSegment.GetDuration() <= mPacketizerInput->mPacketSize);
}
void AudioInputProcessing::ProcessOutputData(MediaTrackGraphImpl* aGraph,
AudioDataValue* aBuffer,
size_t aFrames, TrackRate aRate,
uint32_t aChannels) {
MOZ_ASSERT(aGraph->OnGraphThread());
if (!mEnabled || PassThrough(aGraph)) {
return;
}
if (!mPacketizerOutput ||
mPacketizerOutput->mPacketSize != GetPacketSize(aRate) ||
mPacketizerOutput->mChannels != aChannels) {
// It's ok to drop the audio still in the packetizer here: if this changes,
// we changed devices or something.
mPacketizerOutput = Nothing();
mPacketizerOutput.emplace(GetPacketSize(aRate), aChannels);
}
mPacketizerOutput->Input(aBuffer, aFrames);
while (mPacketizerOutput->PacketsAvailable()) {
uint32_t samplesPerPacket =
mPacketizerOutput->mPacketSize * mPacketizerOutput->mChannels;
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->mPacketSize;
framesPerPacketFarend =
converter.Process(mInputDownmixBuffer, packet, framesPerPacketFarend);
interleavedFarend = mInputDownmixBuffer.Data();
channelCountFarend = MAX_CHANNELS;
deinterleavedPacketDataChannelPointers.SetLength(MAX_CHANNELS);
} else {
interleavedFarend = packet;
channelCountFarend = aChannels;
framesPerPacketFarend = mPacketizerOutput->mPacketSize;
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(MediaTrackGraphImpl* aGraph,
const AudioSegment& aSegment) {
MOZ_ASSERT(!PassThrough(aGraph),
"This should be bypassed when in PassThrough mode.");
MOZ_ASSERT(mEnabled);
MOZ_ASSERT(mPacketizerInput);
MOZ_ASSERT(mPacketizerInput->mPacketSize ==
GetPacketSize(aGraph->GraphRate()));
// Calculate number of the pending frames in mChunksInPacketizer.
auto pendingFrames = [&]() {
TrackTime frames = 0;
for (const auto& p : mChunksInPacketizer) {
frames += p.first;
}
return frames;
};
// Precondition of the Principal-labelling logic below.
MOZ_ASSERT(mPacketizerInput->FramesAvailable() ==
static_cast<uint32_t>(pendingFrames()));
// The WriteToInterleavedBuffer will do upmix or downmix if the channel-count
// in aSegment's chunks is different from mPacketizerInput->mChannels
// WriteToInterleavedBuffer could be avoided once Bug 1729041 is done.
size_t sampleCount = aSegment.WriteToInterleavedBuffer(
mInterleavedBuffer, mPacketizerInput->mChannels);
size_t frameCount =
sampleCount / static_cast<size_t>(mPacketizerInput->mChannels);
// Packetize our input data into 10ms chunks, deinterleave into planar channel
// buffers, process, and append to the right MediaStreamTrack.
mPacketizerInput->Input(mInterleavedBuffer.Elements(),
static_cast<uint32_t>(frameCount));
// Update mChunksInPacketizer and make sure the precondition for the
// Principal-labelling logic still holds.
for (AudioSegment::ConstChunkIterator iter(aSegment); !iter.IsEnded();
iter.Next()) {
MOZ_ASSERT(iter->mDuration > 0);
mChunksInPacketizer.emplace_back(
std::make_pair(iter->mDuration, iter->mPrincipalHandle));
}
MOZ_ASSERT(mPacketizerInput->FramesAvailable() ==
static_cast<uint32_t>(pendingFrames()));
LOG_FRAME(
"(Graph %p, Driver %p) AudioInputProcessing %p Packetizing %zu frames. "
"Packetizer has %u frames (enough for %u packets) now",
aGraph, aGraph->CurrentDriver(), this, frameCount,
mPacketizerInput->FramesAvailable(),
mPacketizerInput->PacketsAvailable());
size_t offset = 0;
while (mPacketizerInput->PacketsAvailable()) {
mPacketCount++;
uint32_t samplesPerPacket =
mPacketizerInput->mPacketSize * mPacketizerInput->mChannels;
if (mInputBuffer.Length() < samplesPerPacket) {
mInputBuffer.SetLength(samplesPerPacket);
}
if (mDeinterleavedBuffer.Length() < samplesPerPacket) {
mDeinterleavedBuffer.SetLength(samplesPerPacket);
}
float* packet = mInputBuffer.Data();
mPacketizerInput->Output(packet);
// Downmix from mPacketizerInput->mChannels to mono if needed. We always
// have floats here, the packetizer performed the conversion. This handles
// sound cards with multiple physical jacks exposed as a single device with
// _n_ discrete channels, where only a single mic is plugged in. Those
// channels are not correlated temporaly since they are discrete channels,
// mixing is just a sum.
AutoTArray<float*, 8> deinterleavedPacketizedInputDataChannelPointers;
uint32_t channelCountInput = 0;
if (mPacketizerInput->mChannels > MAX_CHANNELS) {
channelCountInput = MONO;
deinterleavedPacketizedInputDataChannelPointers.SetLength(
channelCountInput);
deinterleavedPacketizedInputDataChannelPointers[0] =
mDeinterleavedBuffer.Data();
// Downmix to mono (and effectively have a planar buffer) by summing all
// channels in the first channel.
size_t readIndex = 0;
for (size_t i = 0; i < mPacketizerInput->mPacketSize; i++) {
mDeinterleavedBuffer.Data()[i] = 0.;
for (size_t j = 0; j < mPacketizerInput->mChannels; j++) {
mDeinterleavedBuffer.Data()[i] += packet[readIndex++];
}
}
} else {
channelCountInput = mPacketizerInput->mChannels;
// Deinterleave the input data
// Prepare an array pointing to deinterleaved channels.
deinterleavedPacketizedInputDataChannelPointers.SetLength(
channelCountInput);
offset = 0;
for (size_t i = 0;
i < deinterleavedPacketizedInputDataChannelPointers.Length(); ++i) {
deinterleavedPacketizedInputDataChannelPointers[i] =
mDeinterleavedBuffer.Data() + offset;
offset += mPacketizerInput->mPacketSize;
}
// Deinterleave to mInputBuffer, pointed to by inputBufferChannelPointers.
Deinterleave(packet, mPacketizerInput->mPacketSize, channelCountInput,
deinterleavedPacketizedInputDataChannelPointers.Elements());
}
StreamConfig inputConfig(aGraph->GraphRate(), channelCountInput,
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.
CheckedInt<size_t> bufferSize(sizeof(float));
bufferSize *= mPacketizerInput->mPacketSize;
bufferSize *= channelCountInput;
RefPtr<SharedBuffer> buffer = SharedBuffer::Create(bufferSize);
// Prepare channel pointers to the SharedBuffer created above.
AutoTArray<float*, 8> processedOutputChannelPointers;
AutoTArray<const float*, 8> processedOutputChannelPointersConst;
processedOutputChannelPointers.SetLength(channelCountInput);
processedOutputChannelPointersConst.SetLength(channelCountInput);
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->mPacketSize;
}
mAudioProcessing->ProcessStream(
deinterleavedPacketizedInputDataChannelPointers.Elements(), inputConfig,
outputConfig, processedOutputChannelPointers.Elements());
// If logging is enabled, dump the audio processing stats twice a second
if (MOZ_LOG_TEST(gMediaManagerLog, LogLevel::Debug) &&
!(mPacketCount % 50)) {
AudioProcessingStats stats = mAudioProcessing->GetStatistics();
char msg[1024];
size_t offset = 0;
#define AddIfValue(format, member) \
if (stats.member.has_value()) { \
offset += SprintfBuf(msg + offset, sizeof(msg) - offset, \
#member ":" format ", ", stats.member.value()); \
}
AddIfValue("%d", output_rms_dbfs);
AddIfValue("%d", voice_detected);
AddIfValue("%lf", echo_return_loss);
AddIfValue("%lf", echo_return_loss_enhancement);
AddIfValue("%lf", divergent_filter_fraction);
AddIfValue("%d", delay_median_ms);
AddIfValue("%d", delay_standard_deviation_ms);
AddIfValue("%lf", residual_echo_likelihood);
AddIfValue("%lf", residual_echo_likelihood_recent_max);
AddIfValue("%d", delay_ms);
#undef AddIfValue
LOG("AudioProcessing statistics: %s", msg);
}