/*
 *  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/base/refcount.h"
#include "webrtc/modules/include/module_common_types.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.
    };

    struct AudioFrameWithInfo {
      AudioFrame* audio_frame;
      AudioFrameInfo audio_frame_info;
    };

    // The implementation of GetAudioFrameWithInfo should update
    // audio_frame with new audio every time it's called. Implementing
    // classes are allowed to return the same AudioFrame pointer on
    // different calls. The pointer must stay valid until the next
    // mixing call or until this audio source is disconnected from the
    // mixer. The mixer may modify the contents of the passed
    // AudioFrame pointer at any time until the next call to
    // GetAudioFrameWithInfo, or until the source is removed from the
    // mixer.
    virtual AudioFrameWithInfo GetAudioFrameWithInfo(int sample_rate_hz) = 0;

    // A way for a mixer implementation to distinguish participants.
    virtual int Ssrc() = 0;

   protected:
    virtual ~Source() {}
  };

  // Since the mixer is reference counted, the destructor may be
  // called from any thread.
  ~AudioMixer() override {}

  // Returns true if adding/removing was successful. A source is never
  // added twice and removal is never attempted if a source has not
  // been successfully added to the mixer. Addition and removal can
  // happen on different threads.
  virtual bool AddSource(Source* audio_source) = 0;
  virtual bool RemoveSource(Source* audio_source) = 0;

  // Performs mixing by asking registered audio sources for audio. The
  // mixed result is placed in the provided AudioFrame. Will only be
  // called from a single thread. The rate and channels arguments
  // specify the rate and number of channels of the mix result.
  virtual void Mix(int sample_rate_hz,
                   size_t number_of_channels,
                   AudioFrame* audio_frame_for_mixing) = 0;
};
}  // namespace webrtc

#endif  // WEBRTC_API_AUDIO_AUDIO_MIXER_H_