audio_mixer_impl.h

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/*

 *  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.

*/

#ifndef MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_

#define MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_

#include <stddef.h>

#include <memory>

#include <vector>

#include "api/array_view.h"

#include "api/audio/audio_frame.h"

#include "api/audio/audio_mixer.h"

#include "api/scoped_refptr.h"

#include "modules/audio_mixer/frame_combiner.h"

#include "modules/audio_mixer/output_rate_calculator.h"

#include "rtc_base/race_checker.h"

#include "rtc_base/synchronization/mutex.h"

#include "rtc_base/thread_annotations.h"

namespace webrtc {

class AudioMixerImpl : public AudioMixer {

 public:

  struct SourceStatus;

  // AudioProcessing only accepts 10 ms frames.

  static const int kFrameDurationInMs = 10;

  static rtc::scoped_refptr<AudioMixerImpl> Create();

  static rtc::scoped_refptr<AudioMixerImpl> Create(

      std::unique_ptr<OutputRateCalculator> output_rate_calculator,

      bool use_limiter);

  ~AudioMixerImpl() override;

  AudioMixerImpl(const AudioMixerImpl&) = delete;

  AudioMixerImpl& operator=(const AudioMixerImpl&) = delete;

  // AudioMixer functions

  bool AddSource(Source* audio_source) override;

  void RemoveSource(Source* audio_source) override;

  void Mix(size_t number_of_channels,

           AudioFrame* audio_frame_for_mixing) override

      RTC_LOCKS_EXCLUDED(mutex_);

 protected:

  AudioMixerImpl(std::unique_ptr<OutputRateCalculator> output_rate_calculator,

                 bool use_limiter);

 private:

  struct HelperContainers;

  void UpdateSourceCountStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);

  // Fetches audio frames to mix from sources.

  rtc::ArrayView<AudioFrame* const> GetAudioFromSources(int output_frequency)

      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);

  // The critical section lock guards audio source insertion and

  // removal, which can be done from any thread. The race checker

  // checks that mixing is done sequentially.

  mutable Mutex mutex_;

  std::unique_ptr<OutputRateCalculator> output_rate_calculator_;

  // List of all audio sources.

  std::vector<std::unique_ptr<SourceStatus>> audio_source_list_

      RTC_GUARDED_BY(mutex_);

  const std::unique_ptr<HelperContainers> helper_containers_

      RTC_GUARDED_BY(mutex_);

  // Component that handles actual adding of audio frames.

  FrameCombiner frame_combiner_;

  // The highest source count this mixer has ever had. Used for UMA stats.

  size_t max_source_count_ever_ = 0;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_