You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
81 lines
3.0 KiB
81 lines
3.0 KiB
/*
|
|
* 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/ref_count.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_
|