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/*
* 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 API_AUDIO_AUDIO_MIXER_H_
#define API_AUDIO_AUDIO_MIXER_H_
#include <memory>
#include "api/audio/audio_frame.h"
#include "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 // API_AUDIO_AUDIO_MIXER_H_
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