webrtc_m130/modules/audio_mixer/audio_mixer_impl.cc

/*
 *  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 "modules/audio_mixer/audio_mixer_impl.h"

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

#include "modules/audio_mixer/audio_frame_manipulator.h"
#include "modules/audio_mixer/default_output_rate_calculator.h"
#include "rtc_base/logging.h"
#include "rtc_base/refcountedobject.h"
#include "system_wrappers/include/field_trial.h"

namespace webrtc {
namespace {

struct SourceFrame {
  SourceFrame(AudioMixerImpl::SourceStatus* source_status,
              AudioFrame* audio_frame,
              bool muted)
      : source_status(source_status), audio_frame(audio_frame), muted(muted) {
    RTC_DCHECK(source_status);
    RTC_DCHECK(audio_frame);
    if (!muted) {
      energy = AudioMixerCalculateEnergy(*audio_frame);
    }
  }

  SourceFrame(AudioMixerImpl::SourceStatus* source_status,
              AudioFrame* audio_frame,
              bool muted,
              uint32_t energy)
      : source_status(source_status),
        audio_frame(audio_frame),
        muted(muted),
        energy(energy) {
    RTC_DCHECK(source_status);
    RTC_DCHECK(audio_frame);
  }

  AudioMixerImpl::SourceStatus* source_status = nullptr;
  AudioFrame* audio_frame = nullptr;
  bool muted = true;
  uint32_t energy = 0;
};

// ShouldMixBefore(a, b) is used to select mixer sources.
bool ShouldMixBefore(const SourceFrame& a, const SourceFrame& b) {
  if (a.muted != b.muted) {
    return b.muted;
  }

  const auto a_activity = a.audio_frame->vad_activity_;
  const auto b_activity = b.audio_frame->vad_activity_;

  if (a_activity != b_activity) {
    return a_activity == AudioFrame::kVadActive;
  }

  return a.energy > b.energy;
}

void RampAndUpdateGain(
    const std::vector<SourceFrame>& mixed_sources_and_frames) {
  for (const auto& source_frame : mixed_sources_and_frames) {
    float target_gain = source_frame.source_status->is_mixed ? 1.0f : 0.0f;
    Ramp(source_frame.source_status->gain, target_gain,
         source_frame.audio_frame);
    source_frame.source_status->gain = target_gain;
  }
}

AudioMixerImpl::SourceStatusList::const_iterator FindSourceInList(
    AudioMixerImpl::Source const* audio_source,
    AudioMixerImpl::SourceStatusList const* audio_source_list) {
  return std::find_if(
      audio_source_list->begin(), audio_source_list->end(),
      [audio_source](const std::unique_ptr<AudioMixerImpl::SourceStatus>& p) {
        return p->audio_source == audio_source;
      });
}

FrameCombiner::LimiterType ChooseLimiterType(bool use_limiter) {
  using LimiterType = FrameCombiner::LimiterType;
  if (!use_limiter) {
    return LimiterType::kNoLimiter;
  } else if (field_trial::IsEnabled("WebRTC-ApmGainController2Limiter")) {
    return LimiterType::kApmAgc2Limiter;
  } else {
    return LimiterType::kApmAgcLimiter;
  }
}
}  // namespace

AudioMixerImpl::AudioMixerImpl(
    std::unique_ptr<OutputRateCalculator> output_rate_calculator,
    bool use_limiter)
    : output_rate_calculator_(std::move(output_rate_calculator)),
      output_frequency_(0),
      sample_size_(0),
      audio_source_list_(),
      frame_combiner_(ChooseLimiterType(use_limiter)) {}

AudioMixerImpl::~AudioMixerImpl() {}

rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create() {
  return Create(std::unique_ptr<DefaultOutputRateCalculator>(
                    new DefaultOutputRateCalculator()),
                true);
}

rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
    std::unique_ptr<OutputRateCalculator> output_rate_calculator,
    bool use_limiter) {
  return rtc::scoped_refptr<AudioMixerImpl>(
      new rtc::RefCountedObject<AudioMixerImpl>(
          std::move(output_rate_calculator), use_limiter));
}

void AudioMixerImpl::Mix(size_t number_of_channels,
                         AudioFrame* audio_frame_for_mixing) {
  RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2);
  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);

  CalculateOutputFrequency();

  {
    rtc::CritScope lock(&crit_);
    const size_t number_of_streams = audio_source_list_.size();
    frame_combiner_.Combine(GetAudioFromSources(), number_of_channels,
                            OutputFrequency(), number_of_streams,
                            audio_frame_for_mixing);
  }

  return;
}

void AudioMixerImpl::CalculateOutputFrequency() {
  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
  rtc::CritScope lock(&crit_);

  std::vector<int> preferred_rates;
  std::transform(audio_source_list_.begin(), audio_source_list_.end(),
                 std::back_inserter(preferred_rates),
                 [&](std::unique_ptr<SourceStatus>& a) {
                   return a->audio_source->PreferredSampleRate();
                 });

  output_frequency_ =
      output_rate_calculator_->CalculateOutputRate(preferred_rates);
  sample_size_ = (output_frequency_ * kFrameDurationInMs) / 1000;
}

int AudioMixerImpl::OutputFrequency() const {
  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
  return output_frequency_;
}

bool AudioMixerImpl::AddSource(Source* audio_source) {
  RTC_DCHECK(audio_source);
  rtc::CritScope lock(&crit_);
  RTC_DCHECK(FindSourceInList(audio_source, &audio_source_list_) ==
             audio_source_list_.end())
      << "Source already added to mixer";
  audio_source_list_.emplace_back(new SourceStatus(audio_source, false, 0));
  return true;
}

void AudioMixerImpl::RemoveSource(Source* audio_source) {
  RTC_DCHECK(audio_source);
  rtc::CritScope lock(&crit_);
  const auto iter = FindSourceInList(audio_source, &audio_source_list_);
  RTC_DCHECK(iter != audio_source_list_.end()) << "Source not present in mixer";
  audio_source_list_.erase(iter);
}

AudioFrameList AudioMixerImpl::GetAudioFromSources() {
  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
  AudioFrameList result;
  std::vector<SourceFrame> audio_source_mixing_data_list;
  std::vector<SourceFrame> ramp_list;

  // Get audio from the audio sources and put it in the SourceFrame vector.
  for (auto& source_and_status : audio_source_list_) {
    const auto audio_frame_info =
        source_and_status->audio_source->GetAudioFrameWithInfo(
            OutputFrequency(), &source_and_status->audio_frame);

    if (audio_frame_info == Source::AudioFrameInfo::kError) {
      RTC_LOG_F(LS_WARNING) << "failed to GetAudioFrameWithInfo() from source";
      continue;
    }
    audio_source_mixing_data_list.emplace_back(
        source_and_status.get(), &source_and_status->audio_frame,
        audio_frame_info == Source::AudioFrameInfo::kMuted);
  }

  // Sort frames by sorting function.
  std::sort(audio_source_mixing_data_list.begin(),
            audio_source_mixing_data_list.end(), ShouldMixBefore);

  int max_audio_frame_counter = kMaximumAmountOfMixedAudioSources;

  // Go through list in order and put unmuted frames in result list.
  for (const auto& p : audio_source_mixing_data_list) {
    // Filter muted.
    if (p.muted) {
      p.source_status->is_mixed = 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.source_status, p.audio_frame, false, -1);
      is_mixed = true;
    }
    p.source_status->is_mixed = is_mixed;
  }
  RampAndUpdateGain(ramp_list);
  return result;
}

bool AudioMixerImpl::GetAudioSourceMixabilityStatusForTest(
    AudioMixerImpl::Source* audio_source) const {
  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
  rtc::CritScope lock(&crit_);

  const auto iter = FindSourceInList(audio_source, &audio_source_list_);
  if (iter != audio_source_list_.end()) {
    return (*iter)->is_mixed;
  }

  RTC_LOG(LS_ERROR) << "Audio source unknown";
  return false;
}
}  // namespace webrtc