webrtc/modules/audio_mixer/audio_mixer_impl.cc - src - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"

 #include <algorithm>
 #include <functional>
 #include <utility>

 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"

 namespace webrtc {
 namespace {

 class SourceFrame {
  public:
   SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before)
       : audio_source_(p),
         audio_frame_(a),
         muted_(m),
         was_mixed_before_(was_mixed_before) {
     if (!muted_) {
       energy_ = NewMixerCalculateEnergy(*a);
     }
   }

   SourceFrame(MixerAudioSource* p,
               AudioFrame* a,
               bool m,
               bool was_mixed_before,
               uint32_t energy)
       : audio_source_(p),
         audio_frame_(a),
         muted_(m),
         energy_(energy),
         was_mixed_before_(was_mixed_before) {}

   // a.shouldMixBefore(b) is used to select mixer participants.
   bool shouldMixBefore(const SourceFrame& other) const {
     if (muted_ != other.muted_) {
       return other.muted_;
     }

     const auto our_activity = audio_frame_->vad_activity_;
     const auto other_activity = other.audio_frame_->vad_activity_;

     if (our_activity != other_activity) {
       return our_activity == AudioFrame::kVadActive;
     }

     return energy_ > other.energy_;
   }

   MixerAudioSource* audio_source_;
   AudioFrame* audio_frame_;
   bool muted_;
   uint32_t energy_;
   bool was_mixed_before_;
 };

 // Remixes a frame between stereo and mono.
 void RemixFrame(AudioFrame* frame, size_t number_of_channels) {
   RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2);
   if (frame->num_channels_ == 1 && number_of_channels == 2) {
     AudioFrameOperations::MonoToStereo(frame);
   } else if (frame->num_channels_ == 2 && number_of_channels == 1) {
     AudioFrameOperations::StereoToMono(frame);
   }
 }

 void Ramp(const std::vector<SourceFrame>& mixed_sources_and_frames) {
   for (const auto& source_frame : mixed_sources_and_frames) {
     // Ramp in previously unmixed.
     if (!source_frame.was_mixed_before_) {
       NewMixerRampIn(source_frame.audio_frame_);
     }

     const bool is_mixed = source_frame.audio_source_->mix_history_->IsMixed();
     // Ramp out currently unmixed.
     if (source_frame.was_mixed_before_ && !is_mixed) {
       NewMixerRampOut(source_frame.audio_frame_);
     }
   }
 }

 }  // namespace

 MixerAudioSource::MixerAudioSource() : mix_history_(new NewMixHistory()) {}

 MixerAudioSource::~MixerAudioSource() {
   delete mix_history_;
 }

 bool MixerAudioSource::IsMixed() const {
   return mix_history_->IsMixed();
 }

 NewMixHistory::NewMixHistory() : is_mixed_(0) {}

 NewMixHistory::~NewMixHistory() {}

 bool NewMixHistory::IsMixed() const {
   return is_mixed_;
 }

 bool NewMixHistory::WasMixed() const {
   // Was mixed is the same as is mixed depending on perspective. This function
   // is for the perspective of NewAudioConferenceMixerImpl.
   return IsMixed();
 }

 int32_t NewMixHistory::SetIsMixed(const bool mixed) {
   is_mixed_ = mixed;
   return 0;
 }

 void NewMixHistory::ResetMixedStatus() {
   is_mixed_ = false;
 }

 std::unique_ptr<AudioMixer> AudioMixer::Create(int id) {
   return AudioMixerImpl::Create(id);
 }

 AudioMixerImpl::AudioMixerImpl(int id, std::unique_ptr<AudioProcessing> limiter)
     : id_(id),
       audio_source_list_(),
       additional_audio_source_list_(),
       num_mixed_audio_sources_(0),
       use_limiter_(true),
       time_stamp_(0),
       limiter_(std::move(limiter)) {
   SetOutputFrequency(kDefaultFrequency);
   thread_checker_.DetachFromThread();
 }

 AudioMixerImpl::~AudioMixerImpl() {}

 std::unique_ptr<AudioMixer> AudioMixerImpl::Create(int id) {
   Config config;
   config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
   std::unique_ptr<AudioProcessing> limiter(AudioProcessing::Create(config));
   if (!limiter.get())
     return nullptr;

   if (limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
       limiter->kNoError)
     return nullptr;

   // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
   // divide-by-2 but -7 is used instead to give a bit of headroom since the
   // AGC is not a hard limiter.
   if (limiter->gain_control()->set_target_level_dbfs(7) != limiter->kNoError)
     return nullptr;

   if (limiter->gain_control()->set_compression_gain_db(0) != limiter->kNoError)
     return nullptr;

   if (limiter->gain_control()->enable_limiter(true) != limiter->kNoError)
     return nullptr;

   if (limiter->gain_control()->Enable(true) != limiter->kNoError)
     return nullptr;

   return std::unique_ptr<AudioMixer>(
       new AudioMixerImpl(id, std::move(limiter)));
 }

 void AudioMixerImpl::Mix(int sample_rate,
                          size_t number_of_channels,
                          AudioFrame* audio_frame_for_mixing) {
   RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2);
   RTC_DCHECK_RUN_ON(&thread_checker_);

   if (sample_rate != kNbInHz && sample_rate != kWbInHz &&
       sample_rate != kSwbInHz && sample_rate != kFbInHz) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
                  "Invalid frequency: %d", sample_rate);
     RTC_NOTREACHED();
     return;
   }

   if (OutputFrequency() != sample_rate) {
     SetOutputFrequency(static_cast<Frequency>(sample_rate));
   }

   AudioFrameList mix_list;
   AudioFrameList anonymous_mix_list;
   int num_mixed_audio_sources;
   {
     rtc::CritScope lock(&crit_);
     mix_list = GetNonAnonymousAudio();
     anonymous_mix_list = GetAnonymousAudio();
     num_mixed_audio_sources = static_cast<int>(num_mixed_audio_sources_);
   }

   mix_list.insert(mix_list.begin(), anonymous_mix_list.begin(),
                   anonymous_mix_list.end());

   for (const auto& frame : mix_list) {
     RemixFrame(frame, number_of_channels);
   }

   audio_frame_for_mixing->UpdateFrame(
       -1, time_stamp_, NULL, 0, OutputFrequency(), AudioFrame::kNormalSpeech,
       AudioFrame::kVadPassive, number_of_channels);

   time_stamp_ += static_cast<uint32_t>(sample_size_);

   use_limiter_ = num_mixed_audio_sources > 1;

   // We only use the limiter if we're actually mixing multiple streams.
   MixFromList(audio_frame_for_mixing, mix_list, id_, use_limiter_);

   if (audio_frame_for_mixing->samples_per_channel_ == 0) {
     // Nothing was mixed, set the audio samples to silence.
     audio_frame_for_mixing->samples_per_channel_ = sample_size_;
     audio_frame_for_mixing->Mute();
   } else {
     // Only call the limiter if we have something to mix.
     LimitMixedAudio(audio_frame_for_mixing);
   }

   // Pass the final result to the level indicator.
   audio_level_.ComputeLevel(*audio_frame_for_mixing);

   return;
 }

 int32_t AudioMixerImpl::SetOutputFrequency(const Frequency& frequency) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   output_frequency_ = frequency;
   sample_size_ =
       static_cast<size_t>((output_frequency_ * kFrameDurationInMs) / 1000);

   return 0;
 }

 AudioMixer::Frequency AudioMixerImpl::OutputFrequency() const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   return output_frequency_;
 }

 int32_t AudioMixerImpl::SetMixabilityStatus(MixerAudioSource* audio_source,
                                             bool mixable) {
   if (!mixable) {
     // Anonymous audio sources are in a separate list. Make sure that the
     // audio source is in the _audioSourceList if it is being mixed.
     SetAnonymousMixabilityStatus(audio_source, false);
   }
   {
     rtc::CritScope lock(&crit_);
     const bool is_mixed =
         IsAudioSourceInList(*audio_source, audio_source_list_);
     // API must be called with a new state.
     if (!(mixable ^ is_mixed)) {
       WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
                    "Mixable is aready %s", is_mixed ? "ON" : "off");
       return -1;
     }
     bool success = false;
     if (mixable) {
       success = AddAudioSourceToList(audio_source, &audio_source_list_);
     } else {
       success = RemoveAudioSourceFromList(audio_source, &audio_source_list_);
     }
     if (!success) {
       WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
                    "failed to %s audio_source", mixable ? "add" : "remove");
       RTC_NOTREACHED();
       return -1;
     }

     size_t num_mixed_non_anonymous = audio_source_list_.size();
     if (num_mixed_non_anonymous > kMaximumAmountOfMixedAudioSources) {
       num_mixed_non_anonymous = kMaximumAmountOfMixedAudioSources;
     }
     num_mixed_audio_sources_ =
         num_mixed_non_anonymous + additional_audio_source_list_.size();
   }
   return 0;
 }

 bool AudioMixerImpl::MixabilityStatus(
     const MixerAudioSource& audio_source) const {
   rtc::CritScope lock(&crit_);
   return IsAudioSourceInList(audio_source, audio_source_list_);
 }

 int32_t AudioMixerImpl::SetAnonymousMixabilityStatus(
     MixerAudioSource* audio_source,
     bool anonymous) {
   rtc::CritScope lock(&crit_);
   if (IsAudioSourceInList(*audio_source, additional_audio_source_list_)) {
     if (anonymous) {
       return 0;
     }
     if (!RemoveAudioSourceFromList(audio_source,
                                    &additional_audio_source_list_)) {
       WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
                    "unable to remove audio_source from anonymous list");
       RTC_NOTREACHED();
       return -1;
     }
     return AddAudioSourceToList(audio_source, &audio_source_list_) ? 0 : -1;
   }
   if (!anonymous) {
     return 0;
   }
   const bool mixable =
       RemoveAudioSourceFromList(audio_source, &audio_source_list_);
   if (!mixable) {
     WEBRTC_TRACE(
         kTraceWarning, kTraceAudioMixerServer, id_,
         "audio_source must be registered before turning it into anonymous");
     // Setting anonymous status is only possible if MixerAudioSource is
     // already registered.
     return -1;
   }
   return AddAudioSourceToList(audio_source, &additional_audio_source_list_)
              ? 0
              : -1;
 }

 bool AudioMixerImpl::AnonymousMixabilityStatus(
     const MixerAudioSource& audio_source) const {
   rtc::CritScope lock(&crit_);
   return IsAudioSourceInList(audio_source, additional_audio_source_list_);
 }

 AudioFrameList AudioMixerImpl::GetNonAnonymousAudio() const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
                "GetNonAnonymousAudio()");
   AudioFrameList result;
   std::vector<SourceFrame> audio_source_mixing_data_list;
   std::vector<SourceFrame> ramp_list;

   // Get audio source audio and put it in the struct vector.
   for (auto* const audio_source : audio_source_list_) {
     auto audio_frame_with_info = audio_source->GetAudioFrameWithMuted(
         id_, static_cast<int>(OutputFrequency()));

     const auto audio_frame_info = audio_frame_with_info.audio_frame_info;
     AudioFrame* audio_source_audio_frame = audio_frame_with_info.audio_frame;

     if (audio_frame_info == MixerAudioSource::AudioFrameInfo::kError) {
       WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
                    "failed to GetAudioFrameWithMuted() from participant");
       continue;
     }
     audio_source_mixing_data_list.emplace_back(
         audio_source, audio_source_audio_frame,
         audio_frame_info == MixerAudioSource::AudioFrameInfo::kMuted,
         audio_source->mix_history_->WasMixed());
   }

   // Sort frames by sorting function.
   std::sort(audio_source_mixing_data_list.begin(),
             audio_source_mixing_data_list.end(),
             std::mem_fn(&SourceFrame::shouldMixBefore));

   int max_audio_frame_counter = kMaximumAmountOfMixedAudioSources;

   // Go through list in order and put unmuted frames in result list.
   for (const SourceFrame& p : audio_source_mixing_data_list) {
     // Filter muted.
     if (p.muted_) {
       p.audio_source_->mix_history_->SetIsMixed(false);
       continue;
     }

     // Add frame to result vector for mixing.
     bool is_mixed = false;
     if (max_audio_frame_counter > 0) {
       --max_audio_frame_counter;
       result.push_back(p.audio_frame_);
       ramp_list.emplace_back(p.audio_source_, p.audio_frame_, false,
                              p.was_mixed_before_, -1);
       is_mixed = true;
     }
     p.audio_source_->mix_history_->SetIsMixed(is_mixed);
   }
   Ramp(ramp_list);
   return result;
 }

 AudioFrameList AudioMixerImpl::GetAnonymousAudio() const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
                "GetAnonymousAudio()");
   // The GetAudioFrameWithMuted() callback may result in the audio source being
   // removed from additionalAudioFramesList_. If that happens it will
   // invalidate any iterators. Create a copy of the audio sources list such
   // that the list of participants can be traversed safely.
   std::vector<SourceFrame> ramp_list;
   MixerAudioSourceList additional_audio_sources_list;
   AudioFrameList result;
   additional_audio_sources_list.insert(additional_audio_sources_list.begin(),
                                        additional_audio_source_list_.begin(),
                                        additional_audio_source_list_.end());

   for (const auto& audio_source : additional_audio_sources_list) {
     const auto audio_frame_with_info =
         audio_source->GetAudioFrameWithMuted(id_, OutputFrequency());
     const auto ret = audio_frame_with_info.audio_frame_info;
     AudioFrame* audio_frame = audio_frame_with_info.audio_frame;
     if (ret == MixerAudioSource::AudioFrameInfo::kError) {
       WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
                    "failed to GetAudioFrameWithMuted() from audio_source");
       continue;
     }
     if (ret != MixerAudioSource::AudioFrameInfo::kMuted) {
       result.push_back(audio_frame);
       ramp_list.emplace_back(audio_source, audio_frame, false,
                              audio_source->mix_history_->IsMixed(), 0);
       audio_source->mix_history_->SetIsMixed(true);
     }
   }
   Ramp(ramp_list);
   return result;
 }

 bool AudioMixerImpl::IsAudioSourceInList(
     const MixerAudioSource& audio_source,
     const MixerAudioSourceList& audio_source_list) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
                "IsAudioSourceInList(audio_source,audio_source_list)");
   return std::find(audio_source_list.begin(), audio_source_list.end(),
                    &audio_source) != audio_source_list.end();
 }

 bool AudioMixerImpl::AddAudioSourceToList(
     MixerAudioSource* audio_source,
     MixerAudioSourceList* audio_source_list) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
                "AddAudioSourceToList(audio_source, audio_source_list)");
   audio_source_list->push_back(audio_source);
   // Make sure that the mixed status is correct for new MixerAudioSource.
   audio_source->mix_history_->ResetMixedStatus();
   return true;
 }

 bool AudioMixerImpl::RemoveAudioSourceFromList(
     MixerAudioSource* audio_source,
     MixerAudioSourceList* audio_source_list) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
                "RemoveAudioSourceFromList(audio_source, audio_source_list)");
   const auto iter = std::find(audio_source_list->begin(),
                               audio_source_list->end(), audio_source);
   if (iter != audio_source_list->end()) {
     audio_source_list->erase(iter);
     // AudioSource is no longer mixed, reset to default.
     audio_source->mix_history_->ResetMixedStatus();
     return true;
   } else {
     return false;
   }
 }

 int32_t AudioMixerImpl::MixFromList(AudioFrame* mixed_audio,
                                     const AudioFrameList& audio_frame_list,
                                     int32_t id,
                                     bool use_limiter) {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id,
                "MixFromList(mixed_audio, audio_frame_list)");
   if (audio_frame_list.empty())
     return 0;

   if (audio_frame_list.size() == 1) {
     mixed_audio->timestamp_ = audio_frame_list.front()->timestamp_;
     mixed_audio->elapsed_time_ms_ = audio_frame_list.front()->elapsed_time_ms_;
   } else {
     // TODO(wu): Issue 3390.
     // Audio frame timestamp is only supported in one channel case.
     mixed_audio->timestamp_ = 0;
     mixed_audio->elapsed_time_ms_ = -1;
   }

   for (const auto& frame : audio_frame_list) {
     RTC_DCHECK_EQ(mixed_audio->sample_rate_hz_, frame->sample_rate_hz_);
     RTC_DCHECK_EQ(
         frame->samples_per_channel_,
         static_cast<size_t>(
             (mixed_audio->sample_rate_hz_ * kFrameDurationInMs) / 1000));

     // Mix |f.frame| into |mixed_audio|, with saturation protection.
     // These effect is applied to |f.frame| itself prior to mixing.
     if (use_limiter) {
       // Divide by two to avoid saturation in the mixing.
       // This is only meaningful if the limiter will be used.
       *frame >>= 1;
     }
     RTC_DCHECK_EQ(frame->num_channels_, mixed_audio->num_channels_);
     *mixed_audio += *frame;
   }
   return 0;
 }

 bool AudioMixerImpl::LimitMixedAudio(AudioFrame* mixed_audio) const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   if (!use_limiter_) {
     return true;
   }

   // Smoothly limit the mixed frame.
   const int error = limiter_->ProcessStream(mixed_audio);

   // And now we can safely restore the level. This procedure results in
   // some loss of resolution, deemed acceptable.
   //
   // It's possible to apply the gain in the AGC (with a target level of 0 dbFS
   // and compression gain of 6 dB). However, in the transition frame when this
   // is enabled (moving from one to two audio sources) it has the potential to
   // create discontinuities in the mixed frame.
   //
   // Instead we double the frame (with addition since left-shifting a
   // negative value is undefined).
   *mixed_audio += *mixed_audio;

   if (error != limiter_->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
                  "Error from AudioProcessing: %d", error);
     RTC_NOTREACHED();
     return false;
   }
   return true;
 }

 int AudioMixerImpl::GetOutputAudioLevel() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   const int level = audio_level_.Level();
   WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
                "GetAudioOutputLevel() => level=%d", level);
   return level;
 }

 int AudioMixerImpl::GetOutputAudioLevelFullRange() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   const int level = audio_level_.LevelFullRange();
   WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
                "GetAudioOutputLevelFullRange() => level=%d", level);
   return level;
 }
 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"

	#include <algorithm>
	#include <functional>
	#include <utility>

	#include "webrtc/base/thread_annotations.h"
	#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
	#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
	#include "webrtc/modules/audio_processing/include/audio_processing.h"
	#include "webrtc/modules/utility/include/audio_frame_operations.h"
	#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
	#include "webrtc/system_wrappers/include/trace.h"

	namespace webrtc {
	namespace {

	class SourceFrame {
	public:
	SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before)
	: audio_source_(p),
	audio_frame_(a),
	muted_(m),
	was_mixed_before_(was_mixed_before) {
	if (!muted_) {
	energy_ = NewMixerCalculateEnergy(*a);
	}
	}

	SourceFrame(MixerAudioSource* p,
	AudioFrame* a,
	bool m,
	bool was_mixed_before,
	uint32_t energy)
	: audio_source_(p),
	audio_frame_(a),
	muted_(m),
	energy_(energy),
	was_mixed_before_(was_mixed_before) {}

	// a.shouldMixBefore(b) is used to select mixer participants.
	bool shouldMixBefore(const SourceFrame& other) const {
	if (muted_ != other.muted_) {
	return other.muted_;
	}

	const auto our_activity = audio_frame_->vad_activity_;
	const auto other_activity = other.audio_frame_->vad_activity_;

	if (our_activity != other_activity) {
	return our_activity == AudioFrame::kVadActive;
	}

	return energy_ > other.energy_;
	}

	MixerAudioSource* audio_source_;
	AudioFrame* audio_frame_;
	bool muted_;
	uint32_t energy_;
	bool was_mixed_before_;
	};

	// Remixes a frame between stereo and mono.
	void RemixFrame(AudioFrame* frame, size_t number_of_channels) {
	RTC_DCHECK(number_of_channels == 1 \|\| number_of_channels == 2);
	if (frame->num_channels_ == 1 && number_of_channels == 2) {
	AudioFrameOperations::MonoToStereo(frame);
	} else if (frame->num_channels_ == 2 && number_of_channels == 1) {
	AudioFrameOperations::StereoToMono(frame);
	}
	}

	void Ramp(const std::vector<SourceFrame>& mixed_sources_and_frames) {
	for (const auto& source_frame : mixed_sources_and_frames) {
	// Ramp in previously unmixed.
	if (!source_frame.was_mixed_before_) {
	NewMixerRampIn(source_frame.audio_frame_);
	}

	const bool is_mixed = source_frame.audio_source_->mix_history_->IsMixed();
	// Ramp out currently unmixed.
	if (source_frame.was_mixed_before_ && !is_mixed) {
	NewMixerRampOut(source_frame.audio_frame_);
	}
	}
	}

	} // namespace

	MixerAudioSource::MixerAudioSource() : mix_history_(new NewMixHistory()) {}

	MixerAudioSource::~MixerAudioSource() {
	delete mix_history_;
	}

	bool MixerAudioSource::IsMixed() const {
	return mix_history_->IsMixed();
	}

	NewMixHistory::NewMixHistory() : is_mixed_(0) {}

	NewMixHistory::~NewMixHistory() {}

	bool NewMixHistory::IsMixed() const {
	return is_mixed_;
	}

	bool NewMixHistory::WasMixed() const {
	// Was mixed is the same as is mixed depending on perspective. This function
	// is for the perspective of NewAudioConferenceMixerImpl.
	return IsMixed();
	}

	int32_t NewMixHistory::SetIsMixed(const bool mixed) {
	is_mixed_ = mixed;
	return 0;
	}

	void NewMixHistory::ResetMixedStatus() {
	is_mixed_ = false;
	}

	std::unique_ptr<AudioMixer> AudioMixer::Create(int id) {
	return AudioMixerImpl::Create(id);
	}

	AudioMixerImpl::AudioMixerImpl(int id, std::unique_ptr<AudioProcessing> limiter)
	: id_(id),
	audio_source_list_(),
	additional_audio_source_list_(),
	num_mixed_audio_sources_(0),
	use_limiter_(true),
	time_stamp_(0),
	limiter_(std::move(limiter)) {
	SetOutputFrequency(kDefaultFrequency);
	thread_checker_.DetachFromThread();
	}

	AudioMixerImpl::~AudioMixerImpl() {}

	std::unique_ptr<AudioMixer> AudioMixerImpl::Create(int id) {
	Config config;
	config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
	std::unique_ptr<AudioProcessing> limiter(AudioProcessing::Create(config));
	if (!limiter.get())
	return nullptr;

	if (limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
	limiter->kNoError)
	return nullptr;

	// We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
	// divide-by-2 but -7 is used instead to give a bit of headroom since the
	// AGC is not a hard limiter.
	if (limiter->gain_control()->set_target_level_dbfs(7) != limiter->kNoError)
	return nullptr;

	if (limiter->gain_control()->set_compression_gain_db(0) != limiter->kNoError)
	return nullptr;

	if (limiter->gain_control()->enable_limiter(true) != limiter->kNoError)
	return nullptr;

	if (limiter->gain_control()->Enable(true) != limiter->kNoError)
	return nullptr;

	return std::unique_ptr<AudioMixer>(
	new AudioMixerImpl(id, std::move(limiter)));
	}

	void AudioMixerImpl::Mix(int sample_rate,
	size_t number_of_channels,
	AudioFrame* audio_frame_for_mixing) {
	RTC_DCHECK(number_of_channels == 1 \|\| number_of_channels == 2);
	RTC_DCHECK_RUN_ON(&thread_checker_);

	if (sample_rate != kNbInHz && sample_rate != kWbInHz &&
	sample_rate != kSwbInHz && sample_rate != kFbInHz) {
	WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
	"Invalid frequency: %d", sample_rate);
	RTC_NOTREACHED();
	return;
	}

	if (OutputFrequency() != sample_rate) {
	SetOutputFrequency(static_cast<Frequency>(sample_rate));
	}

	AudioFrameList mix_list;
	AudioFrameList anonymous_mix_list;
	int num_mixed_audio_sources;
	{
	rtc::CritScope lock(&crit_);
	mix_list = GetNonAnonymousAudio();
	anonymous_mix_list = GetAnonymousAudio();
	num_mixed_audio_sources = static_cast<int>(num_mixed_audio_sources_);
	}

	mix_list.insert(mix_list.begin(), anonymous_mix_list.begin(),
	anonymous_mix_list.end());

	for (const auto& frame : mix_list) {
	RemixFrame(frame, number_of_channels);
	}

	audio_frame_for_mixing->UpdateFrame(
	-1, time_stamp_, NULL, 0, OutputFrequency(), AudioFrame::kNormalSpeech,
	AudioFrame::kVadPassive, number_of_channels);

	time_stamp_ += static_cast<uint32_t>(sample_size_);

	use_limiter_ = num_mixed_audio_sources > 1;

	// We only use the limiter if we're actually mixing multiple streams.
	MixFromList(audio_frame_for_mixing, mix_list, id_, use_limiter_);

	if (audio_frame_for_mixing->samples_per_channel_ == 0) {
	// Nothing was mixed, set the audio samples to silence.
	audio_frame_for_mixing->samples_per_channel_ = sample_size_;
	audio_frame_for_mixing->Mute();
	} else {
	// Only call the limiter if we have something to mix.
	LimitMixedAudio(audio_frame_for_mixing);
	}

	// Pass the final result to the level indicator.
	audio_level_.ComputeLevel(*audio_frame_for_mixing);

	return;
	}

	int32_t AudioMixerImpl::SetOutputFrequency(const Frequency& frequency) {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	output_frequency_ = frequency;
	sample_size_ =
	static_cast<size_t>((output_frequency_ * kFrameDurationInMs) / 1000);

	return 0;
	}

	AudioMixer::Frequency AudioMixerImpl::OutputFrequency() const {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	return output_frequency_;
	}

	int32_t AudioMixerImpl::SetMixabilityStatus(MixerAudioSource* audio_source,
	bool mixable) {
	if (!mixable) {
	// Anonymous audio sources are in a separate list. Make sure that the
	// audio source is in the _audioSourceList if it is being mixed.
	SetAnonymousMixabilityStatus(audio_source, false);
	}
	{
	rtc::CritScope lock(&crit_);
	const bool is_mixed =
	IsAudioSourceInList(*audio_source, audio_source_list_);
	// API must be called with a new state.
	if (!(mixable ^ is_mixed)) {
	WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
	"Mixable is aready %s", is_mixed ? "ON" : "off");
	return -1;
	}
	bool success = false;
	if (mixable) {
	success = AddAudioSourceToList(audio_source, &audio_source_list_);
	} else {
	success = RemoveAudioSourceFromList(audio_source, &audio_source_list_);
	}
	if (!success) {
	WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
	"failed to %s audio_source", mixable ? "add" : "remove");
	RTC_NOTREACHED();
	return -1;
	}

	size_t num_mixed_non_anonymous = audio_source_list_.size();
	if (num_mixed_non_anonymous > kMaximumAmountOfMixedAudioSources) {
	num_mixed_non_anonymous = kMaximumAmountOfMixedAudioSources;
	}
	num_mixed_audio_sources_ =
	num_mixed_non_anonymous + additional_audio_source_list_.size();
	}
	return 0;
	}

	bool AudioMixerImpl::MixabilityStatus(
	const MixerAudioSource& audio_source) const {
	rtc::CritScope lock(&crit_);
	return IsAudioSourceInList(audio_source, audio_source_list_);
	}

	int32_t AudioMixerImpl::SetAnonymousMixabilityStatus(
	MixerAudioSource* audio_source,
	bool anonymous) {
	rtc::CritScope lock(&crit_);
	if (IsAudioSourceInList(*audio_source, additional_audio_source_list_)) {
	if (anonymous) {
	return 0;
	}
	if (!RemoveAudioSourceFromList(audio_source,
	&additional_audio_source_list_)) {
	WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
	"unable to remove audio_source from anonymous list");
	RTC_NOTREACHED();
	return -1;
	}
	return AddAudioSourceToList(audio_source, &audio_source_list_) ? 0 : -1;
	}
	if (!anonymous) {
	return 0;
	}
	const bool mixable =
	RemoveAudioSourceFromList(audio_source, &audio_source_list_);
	if (!mixable) {
	WEBRTC_TRACE(
	kTraceWarning, kTraceAudioMixerServer, id_,
	"audio_source must be registered before turning it into anonymous");
	// Setting anonymous status is only possible if MixerAudioSource is
	// already registered.
	return -1;
	}
	return AddAudioSourceToList(audio_source, &additional_audio_source_list_)
	? 0
	: -1;
	}

	bool AudioMixerImpl::AnonymousMixabilityStatus(
	const MixerAudioSource& audio_source) const {
	rtc::CritScope lock(&crit_);
	return IsAudioSourceInList(audio_source, additional_audio_source_list_);
	}

	AudioFrameList AudioMixerImpl::GetNonAnonymousAudio() const {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
	"GetNonAnonymousAudio()");
	AudioFrameList result;
	std::vector<SourceFrame> audio_source_mixing_data_list;
	std::vector<SourceFrame> ramp_list;

	// Get audio source audio and put it in the struct vector.
	for (auto* const audio_source : audio_source_list_) {
	auto audio_frame_with_info = audio_source->GetAudioFrameWithMuted(
	id_, static_cast<int>(OutputFrequency()));

	const auto audio_frame_info = audio_frame_with_info.audio_frame_info;
	AudioFrame* audio_source_audio_frame = audio_frame_with_info.audio_frame;

	if (audio_frame_info == MixerAudioSource::AudioFrameInfo::kError) {
	WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
	"failed to GetAudioFrameWithMuted() from participant");
	continue;
	}
	audio_source_mixing_data_list.emplace_back(
	audio_source, audio_source_audio_frame,
	audio_frame_info == MixerAudioSource::AudioFrameInfo::kMuted,
	audio_source->mix_history_->WasMixed());
	}

	// Sort frames by sorting function.
	std::sort(audio_source_mixing_data_list.begin(),
	audio_source_mixing_data_list.end(),
	std::mem_fn(&SourceFrame::shouldMixBefore));

	int max_audio_frame_counter = kMaximumAmountOfMixedAudioSources;

	// Go through list in order and put unmuted frames in result list.
	for (const SourceFrame& p : audio_source_mixing_data_list) {
	// Filter muted.
	if (p.muted_) {
	p.audio_source_->mix_history_->SetIsMixed(false);
	continue;
	}

	// Add frame to result vector for mixing.
	bool is_mixed = false;
	if (max_audio_frame_counter > 0) {
	--max_audio_frame_counter;
	result.push_back(p.audio_frame_);
	ramp_list.emplace_back(p.audio_source_, p.audio_frame_, false,
	p.was_mixed_before_, -1);
	is_mixed = true;
	}
	p.audio_source_->mix_history_->SetIsMixed(is_mixed);
	}
	Ramp(ramp_list);
	return result;
	}

	AudioFrameList AudioMixerImpl::GetAnonymousAudio() const {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
	"GetAnonymousAudio()");
	// The GetAudioFrameWithMuted() callback may result in the audio source being
	// removed from additionalAudioFramesList_. If that happens it will
	// invalidate any iterators. Create a copy of the audio sources list such
	// that the list of participants can be traversed safely.
	std::vector<SourceFrame> ramp_list;
	MixerAudioSourceList additional_audio_sources_list;
	AudioFrameList result;
	additional_audio_sources_list.insert(additional_audio_sources_list.begin(),
	additional_audio_source_list_.begin(),
	additional_audio_source_list_.end());

	for (const auto& audio_source : additional_audio_sources_list) {
	const auto audio_frame_with_info =
	audio_source->GetAudioFrameWithMuted(id_, OutputFrequency());
	const auto ret = audio_frame_with_info.audio_frame_info;
	AudioFrame* audio_frame = audio_frame_with_info.audio_frame;
	if (ret == MixerAudioSource::AudioFrameInfo::kError) {
	WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
	"failed to GetAudioFrameWithMuted() from audio_source");
	continue;
	}
	if (ret != MixerAudioSource::AudioFrameInfo::kMuted) {
	result.push_back(audio_frame);
	ramp_list.emplace_back(audio_source, audio_frame, false,
	audio_source->mix_history_->IsMixed(), 0);
	audio_source->mix_history_->SetIsMixed(true);
	}
	}
	Ramp(ramp_list);
	return result;
	}

	bool AudioMixerImpl::IsAudioSourceInList(
	const MixerAudioSource& audio_source,
	const MixerAudioSourceList& audio_source_list) const {
	WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
	"IsAudioSourceInList(audio_source,audio_source_list)");
	return std::find(audio_source_list.begin(), audio_source_list.end(),
	&audio_source) != audio_source_list.end();
	}

	bool AudioMixerImpl::AddAudioSourceToList(
	MixerAudioSource* audio_source,
	MixerAudioSourceList* audio_source_list) const {
	WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
	"AddAudioSourceToList(audio_source, audio_source_list)");
	audio_source_list->push_back(audio_source);
	// Make sure that the mixed status is correct for new MixerAudioSource.
	audio_source->mix_history_->ResetMixedStatus();
	return true;
	}

	bool AudioMixerImpl::RemoveAudioSourceFromList(
	MixerAudioSource* audio_source,
	MixerAudioSourceList* audio_source_list) const {
	WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
	"RemoveAudioSourceFromList(audio_source, audio_source_list)");
	const auto iter = std::find(audio_source_list->begin(),
	audio_source_list->end(), audio_source);
	if (iter != audio_source_list->end()) {
	audio_source_list->erase(iter);
	// AudioSource is no longer mixed, reset to default.
	audio_source->mix_history_->ResetMixedStatus();
	return true;
	} else {
	return false;
	}
	}

	int32_t AudioMixerImpl::MixFromList(AudioFrame* mixed_audio,
	const AudioFrameList& audio_frame_list,
	int32_t id,
	bool use_limiter) {
	WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id,
	"MixFromList(mixed_audio, audio_frame_list)");
	if (audio_frame_list.empty())
	return 0;

	if (audio_frame_list.size() == 1) {
	mixed_audio->timestamp_ = audio_frame_list.front()->timestamp_;
	mixed_audio->elapsed_time_ms_ = audio_frame_list.front()->elapsed_time_ms_;
	} else {
	// TODO(wu): Issue 3390.
	// Audio frame timestamp is only supported in one channel case.
	mixed_audio->timestamp_ = 0;
	mixed_audio->elapsed_time_ms_ = -1;
	}

	for (const auto& frame : audio_frame_list) {
	RTC_DCHECK_EQ(mixed_audio->sample_rate_hz_, frame->sample_rate_hz_);
	RTC_DCHECK_EQ(
	frame->samples_per_channel_,
	static_cast<size_t>(
	(mixed_audio->sample_rate_hz_ * kFrameDurationInMs) / 1000));

	// Mix \|f.frame\| into \|mixed_audio\|, with saturation protection.
	// These effect is applied to \|f.frame\| itself prior to mixing.
	if (use_limiter) {
	// Divide by two to avoid saturation in the mixing.
	// This is only meaningful if the limiter will be used.
	*frame >>= 1;
	}
	RTC_DCHECK_EQ(frame->num_channels_, mixed_audio->num_channels_);
	mixed_audio += frame;
	}
	return 0;
	}

	bool AudioMixerImpl::LimitMixedAudio(AudioFrame* mixed_audio) const {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	if (!use_limiter_) {
	return true;
	}

	// Smoothly limit the mixed frame.
	const int error = limiter_->ProcessStream(mixed_audio);

	// And now we can safely restore the level. This procedure results in
	// some loss of resolution, deemed acceptable.
	//
	// It's possible to apply the gain in the AGC (with a target level of 0 dbFS
	// and compression gain of 6 dB). However, in the transition frame when this
	// is enabled (moving from one to two audio sources) it has the potential to
	// create discontinuities in the mixed frame.
	//
	// Instead we double the frame (with addition since left-shifting a
	// negative value is undefined).
	mixed_audio += mixed_audio;

	if (error != limiter_->kNoError) {
	WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
	"Error from AudioProcessing: %d", error);
	RTC_NOTREACHED();
	return false;
	}
	return true;
	}

	int AudioMixerImpl::GetOutputAudioLevel() {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	const int level = audio_level_.Level();
	WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
	"GetAudioOutputLevel() => level=%d", level);
	return level;
	}

	int AudioMixerImpl::GetOutputAudioLevelFullRange() {
	RTC_DCHECK_RUN_ON(&thread_checker_);
	const int level = audio_level_.LevelFullRange();
	WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
	"GetAudioOutputLevelFullRange() => level=%d", level);
	return level;
	}
	} // namespace webrtc