Removals and renamings in the new audio mixer.

webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc

Index: webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
diff --git a/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc b/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
deleted file mode 100644
index 96ce8ce41723b59dd072a22b3d565861adb4acbf..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
+++ /dev/null
@@ -1,599 +0,0 @@
-/*
- *  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/new_audio_conference_mixer_impl.h"
-
-#include <algorithm>
-#include <functional>
-
-#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);
-    }
-  }
-
-  // a.shouldMixBefore(b) is used to select mixer participants.
-  bool shouldMixBefore(const SourceFrame& other) const {
-    if (muted_ != other.muted_) {
-      return other.muted_;
-    }
-
-    auto our_activity = audio_frame_->vad_activity_;
-    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);
-  }
-}
-
-// Mix |frame| into |mixed_frame|, with saturation protection and upmixing.
-// These effects are applied to |frame| itself prior to mixing. Assumes that
-// |mixed_frame| always has at least as many channels as |frame|. Supports
-// stereo at most.
-//
-void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
-  RTC_DCHECK_GE(mixed_frame->num_channels_, frame->num_channels_);
-  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_frame->num_channels_);
-  *mixed_frame += *frame;
-}
-
-}  // namespace
-
-MixerAudioSource::MixerAudioSource() : _mixHistory(new NewMixHistory()) {}
-
-MixerAudioSource::~MixerAudioSource() {
-  delete _mixHistory;
-}
-
-bool MixerAudioSource::IsMixed() const {
-  return _mixHistory->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;
-}
-
-NewAudioConferenceMixer* NewAudioConferenceMixer::Create(int id) {
-  NewAudioConferenceMixerImpl* mixer = new NewAudioConferenceMixerImpl(id);
-  if (!mixer->Init()) {
-    delete mixer;
-    return NULL;
-  }
-  return mixer;
-}
-
-NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
-    : id_(id),
-      output_frequency_(kDefaultFrequency),
-      sample_size_(0),
-      audio_source_list_(),
-      additional_audio_source_list_(),
-      num_mixed_audio_sources_(0),
-      use_limiter_(true),
-      time_stamp_(0) {
-  thread_checker_.DetachFromThread();
-}
-
-NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {}
-
-bool NewAudioConferenceMixerImpl::Init() {
-  crit_.reset(CriticalSectionWrapper::CreateCriticalSection());
-  if (crit_.get() == NULL)
-    return false;
-
-  cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection());
-  if (cb_crit_.get() == NULL)
-    return false;
-
-  Config config;
-  config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
-  limiter_.reset(AudioProcessing::Create(config));
-  if (!limiter_.get())
-    return false;
-
-  if (SetOutputFrequency(kDefaultFrequency) == -1)
-    return false;
-
-  if (limiter_->gain_control()->set_mode(GainControl::kFixedDigital) !=
-      limiter_->kNoError)
-    return false;
-
-  // 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 false;
-
-  if (limiter_->gain_control()->set_compression_gain_db(0) !=
-      limiter_->kNoError)
-    return false;
-
-  if (limiter_->gain_control()->enable_limiter(true) != limiter_->kNoError)
-    return false;
-
-  if (limiter_->gain_control()->Enable(true) != limiter_->kNoError)
-    return false;
-
-  return true;
-}
-
-void NewAudioConferenceMixerImpl::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(thread_checker_.CalledOnValidThread());
-  AudioFrameList mixList;
-  AudioFrameList additionalFramesList;
-  std::map<int, MixerAudioSource*> mixedAudioSourcesMap;
-  {
-    CriticalSectionScoped cs(cb_crit_.get());
-    Frequency mixing_frequency;
-
-    switch (sample_rate) {
-      case 8000:
-        mixing_frequency = kNbInHz;
-        break;
-      case 16000:
-        mixing_frequency = kWbInHz;
-        break;
-      case 32000:
-        mixing_frequency = kSwbInHz;
-        break;
-      case 48000:
-        mixing_frequency = kFbInHz;
-        break;
-      default:
-        RTC_NOTREACHED();
-        return;
-    }
-
-    if (OutputFrequency() != mixing_frequency) {
-      SetOutputFrequency(mixing_frequency);
-    }
-
-    mixList = UpdateToMix(kMaximumAmountOfMixedAudioSources);
-    GetAdditionalAudio(&additionalFramesList);
-  }
-
-  for (FrameAndMuteInfo& frame_and_mute : mixList) {
-    RemixFrame(frame_and_mute.frame, number_of_channels);
-  }
-  for (FrameAndMuteInfo& frame_and_mute : additionalFramesList) {
-    RemixFrame(frame_and_mute.frame, number_of_channels);
-  }
-
-  audio_frame_for_mixing->UpdateFrame(
-      -1, time_stamp_, NULL, 0, output_frequency_, 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 it supports the output sample rate and
-  // we're actually mixing multiple streams.
-  MixFromList(audio_frame_for_mixing, mixList, id_, use_limiter_);
-
-  {
-    CriticalSectionScoped cs(crit_.get());
-    MixAnonomouslyFromList(audio_frame_for_mixing, additionalFramesList);
-
-    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 NewAudioConferenceMixerImpl::SetOutputFrequency(
-    const Frequency& frequency) {
-  CriticalSectionScoped cs(crit_.get());
-
-  output_frequency_ = frequency;
-  sample_size_ =
-      static_cast<size_t>((output_frequency_ * kProcessPeriodicityInMs) / 1000);
-
-  return 0;
-}
-
-NewAudioConferenceMixer::Frequency
-NewAudioConferenceMixerImpl::OutputFrequency() const {
-  CriticalSectionScoped cs(crit_.get());
-  return output_frequency_;
-}
-
-int32_t NewAudioConferenceMixerImpl::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);
-  }
-  size_t numMixedAudioSources;
-  {
-    CriticalSectionScoped cs(cb_crit_.get());
-    const bool isMixed = IsAudioSourceInList(*audio_source, audio_source_list_);
-    // API must be called with a new state.
-    if (!(mixable ^ isMixed)) {
-      WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
-                   "Mixable is aready %s", isMixed ? "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 numMixedNonAnonymous = audio_source_list_.size();
-    if (numMixedNonAnonymous > kMaximumAmountOfMixedAudioSources) {
-      numMixedNonAnonymous = kMaximumAmountOfMixedAudioSources;
-    }
-    numMixedAudioSources =
-        numMixedNonAnonymous + additional_audio_source_list_.size();
-  }
-  // A MixerAudioSource was added or removed. Make sure the scratch
-  // buffer is updated if necessary.
-  // Note: The scratch buffer may only be updated in Process().
-  CriticalSectionScoped cs(crit_.get());
-  num_mixed_audio_sources_ = numMixedAudioSources;
-  return 0;
-}
-
-bool NewAudioConferenceMixerImpl::MixabilityStatus(
-    const MixerAudioSource& audio_source) const {
-  CriticalSectionScoped cs(cb_crit_.get());
-  return IsAudioSourceInList(audio_source, audio_source_list_);
-}
-
-int32_t NewAudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
-    MixerAudioSource* audio_source,
-    bool anonymous) {
-  CriticalSectionScoped cs(cb_crit_.get());
-  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 NewAudioConferenceMixerImpl::AnonymousMixabilityStatus(
-    const MixerAudioSource& audio_source) const {
-  CriticalSectionScoped cs(cb_crit_.get());
-  return IsAudioSourceInList(audio_source, additional_audio_source_list_);
-}
-
-AudioFrameList NewAudioConferenceMixerImpl::UpdateToMix(
-    size_t maxAudioFrameCounter) const {
-  AudioFrameList result;
-  std::vector<SourceFrame> audioSourceMixingDataList;
-
-  // Get audio source audio and put it in the struct vector.
-  for (MixerAudioSource* audio_source : audio_source_list_) {
-    auto audio_frame_with_info = audio_source->GetAudioFrameWithMuted(
-        id_, static_cast<int>(output_frequency_));
-
-    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;
-    }
-    audioSourceMixingDataList.emplace_back(
-        audio_source, audio_source_audio_frame,
-        audio_frame_info == MixerAudioSource::AudioFrameInfo::kMuted,
-        audio_source->_mixHistory->WasMixed());
-  }
-
-  // Sort frames by sorting function.
-  std::sort(audioSourceMixingDataList.begin(), audioSourceMixingDataList.end(),
-            std::mem_fn(&SourceFrame::shouldMixBefore));
-
-  // Go through list in order and put things in mixList.
-  for (SourceFrame& p : audioSourceMixingDataList) {
-    // Filter muted.
-    if (p.muted_) {
-      p.audio_source_->_mixHistory->SetIsMixed(false);
-      continue;
-    }
-
-    // Add frame to result vector for mixing.
-    bool is_mixed = false;
-    if (maxAudioFrameCounter > 0) {
-      --maxAudioFrameCounter;
-      if (!p.was_mixed_before_) {
-        NewMixerRampIn(p.audio_frame_);
-      }
-      result.emplace_back(p.audio_frame_, false);
-      is_mixed = true;
-    }
-
-    // Ramp out unmuted.
-    if (p.was_mixed_before_ && !is_mixed) {
-      NewMixerRampOut(p.audio_frame_);
-      result.emplace_back(p.audio_frame_, false);
-    }
-
-    p.audio_source_->_mixHistory->SetIsMixed(is_mixed);
-  }
-  return result;
-}
-
-void NewAudioConferenceMixerImpl::GetAdditionalAudio(
-    AudioFrameList* additionalFramesList) const {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
-               "GetAdditionalAudio(additionalFramesList)");
-  // 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.
-  MixerAudioSourceList additionalAudioSourceList;
-  additionalAudioSourceList.insert(additionalAudioSourceList.begin(),
-                                   additional_audio_source_list_.begin(),
-                                   additional_audio_source_list_.end());
-
-  for (MixerAudioSourceList::const_iterator audio_source =
-           additionalAudioSourceList.begin();
-       audio_source != additionalAudioSourceList.end(); ++audio_source) {
-    auto audio_frame_with_info =
-        (*audio_source)->GetAudioFrameWithMuted(id_, output_frequency_);
-    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 (audio_frame->samples_per_channel_ == 0) {
-      // Empty frame. Don't use it.
-      continue;
-    }
-    additionalFramesList->push_back(FrameAndMuteInfo(
-        audio_frame, ret == MixerAudioSource::AudioFrameInfo::kMuted));
-  }
-}
-
-bool NewAudioConferenceMixerImpl::IsAudioSourceInList(
-    const MixerAudioSource& audio_source,
-    const MixerAudioSourceList& audioSourceList) const {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
-               "IsAudioSourceInList(audio_source,audioSourceList)");
-  return std::find(audioSourceList.begin(), audioSourceList.end(),
-                   &audio_source) != audioSourceList.end();
-}
-
-bool NewAudioConferenceMixerImpl::AddAudioSourceToList(
-    MixerAudioSource* audio_source,
-    MixerAudioSourceList* audioSourceList) const {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
-               "AddAudioSourceToList(audio_source, audioSourceList)");
-  audioSourceList->push_back(audio_source);
-  // Make sure that the mixed status is correct for new MixerAudioSource.
-  audio_source->_mixHistory->ResetMixedStatus();
-  return true;
-}
-
-bool NewAudioConferenceMixerImpl::RemoveAudioSourceFromList(
-    MixerAudioSource* audio_source,
-    MixerAudioSourceList* audioSourceList) const {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
-               "RemoveAudioSourceFromList(audio_source, audioSourceList)");
-  auto iter =
-      std::find(audioSourceList->begin(), audioSourceList->end(), audio_source);
-  if (iter != audioSourceList->end()) {
-    audioSourceList->erase(iter);
-    // AudioSource is no longer mixed, reset to default.
-    audio_source->_mixHistory->ResetMixedStatus();
-    return true;
-  } else {
-    return false;
-  }
-}
-
-int32_t NewAudioConferenceMixerImpl::MixFromList(
-    AudioFrame* mixedAudio,
-    const AudioFrameList& audioFrameList,
-    int32_t id,
-    bool use_limiter) {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id,
-               "MixFromList(mixedAudio, audioFrameList)");
-  if (audioFrameList.empty())
-    return 0;
-
-  uint32_t position = 0;
-
-  if (audioFrameList.size() == 1) {
-    mixedAudio->timestamp_ = audioFrameList.front().frame->timestamp_;
-    mixedAudio->elapsed_time_ms_ =
-        audioFrameList.front().frame->elapsed_time_ms_;
-  } else {
-    // TODO(wu): Issue 3390.
-    // Audio frame timestamp is only supported in one channel case.
-    mixedAudio->timestamp_ = 0;
-    mixedAudio->elapsed_time_ms_ = -1;
-  }
-
-  for (AudioFrameList::const_iterator iter = audioFrameList.begin();
-       iter != audioFrameList.end(); ++iter) {
-    if (!iter->muted) {
-      MixFrames(mixedAudio, iter->frame, use_limiter);
-    }
-
-    position++;
-  }
-
-  return 0;
-}
-
-// TODO(andrew): consolidate this function with MixFromList.
-int32_t NewAudioConferenceMixerImpl::MixAnonomouslyFromList(
-    AudioFrame* mixedAudio,
-    const AudioFrameList& audioFrameList) const {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
-               "MixAnonomouslyFromList(mixedAudio, audioFrameList)");
-
-  if (audioFrameList.empty())
-    return 0;
-
-  for (AudioFrameList::const_iterator iter = audioFrameList.begin();
-       iter != audioFrameList.end(); ++iter) {
-    if (!iter->muted) {
-      MixFrames(mixedAudio, iter->frame, use_limiter_);
-    }
-  }
-  return 0;
-}
-
-bool NewAudioConferenceMixerImpl::LimitMixedAudio(
-    AudioFrame* mixedAudio) const {
-  if (!use_limiter_) {
-    return true;
-  }
-
-  // Smoothly limit the mixed frame.
-  const int error = limiter_->ProcessStream(mixedAudio);
-
-  // 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).
-  *mixedAudio += *mixedAudio;
-
-  if (error != limiter_->kNoError) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
-                 "Error from AudioProcessing: %d", error);
-    RTC_NOTREACHED();
-    return false;
-  }
-  return true;
-}
-
-int NewAudioConferenceMixerImpl::GetOutputAudioLevel() {
-  const int level = audio_level_.Level();
-  WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
-               "GetAudioOutputLevel() => level=%d", level);
-  return level;
-}
-
-int NewAudioConferenceMixerImpl::GetOutputAudioLevelFullRange() {
-  const int level = audio_level_.LevelFullRange();
-  WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
-               "GetAudioOutputLevelFullRange() => level=%d", level);
-  return level;
-}
-}  // namespace webrtc