text: /*
   *  Copyright (c) 2011 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 WEBRTC_API_AUDIO_AUDIO_MIXER_H_
  #define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
  
  #include <memory>
  
  #include "webrtc/modules/include/module_common_types.h"
  #include "webrtc/rtc_base/refcount.h"
  
  namespace webrtc {
  
  // WORK IN PROGRESS
  // This class is under development and is not yet intended for for use outside
  // of WebRtc/Libjingle.
  class AudioMixer : public rtc::RefCountInterface {
   public:
    // A callback class that all mixer participants must inherit from/implement.
    class Source {
     public:
      enum class AudioFrameInfo {
        kNormal,  // The samples in audio_frame are valid and should be used.
        kMuted,   // The samples in audio_frame should not be used, but
                  // should be implicitly interpreted as zero. Other
                  // fields in audio_frame may be read and should
                  // contain meaningful values.
        kError,   // The audio_frame will not be used.
      };
  
      // Overwrites |audio_frame|. The data_ field is overwritten with
      // 10 ms of new audio (either 1 or 2 interleaved channels) at
      // |sample_rate_hz|. All fields in |audio_frame| must be updated.
      virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
                                                   AudioFrame* audio_frame) = 0;
  
      // A way for a mixer implementation to distinguish participants.
      virtual int Ssrc() const = 0;
  
      // A way for this source to say that GetAudioFrameWithInfo called
      // with this sample rate or higher will not cause quality loss.
      virtual int PreferredSampleRate() const = 0;
  
      virtual ~Source() {}
    };
  
    // Returns true if adding was successful. A source is never added
    // twice. Addition and removal can happen on different threads.
    virtual bool AddSource(Source* audio_source) = 0;
  
    // Removal is never attempted if a source has not been successfully
    // added to the mixer.
    virtual void RemoveSource(Source* audio_source) = 0;
  
    // Performs mixing by asking registered audio sources for audio. The
    // mixed result is placed in the provided AudioFrame. This method
    // will only be called from a single thread. The channels argument
    // specifies the number of channels of the mix result. The mixer
    // should mix at a rate that doesn't cause quality loss of the
    // sources' audio. The mixing rate is one of the rates listed in
    // AudioProcessing::NativeRate. All fields in
    // |audio_frame_for_mixing| must be updated.
    virtual void Mix(size_t number_of_channels,
                     AudioFrame* audio_frame_for_mixing) = 0;
  
   protected:
    // Since the mixer is reference counted, the destructor may be
    // called from any thread.
    ~AudioMixer() override {}
  };
  }  // namespace webrtc
  
  #endif  // WEBRTC_API_AUDIO_AUDIO_MIXER_H_
  