audio_mixer.h (3106B)
1 /* 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef API_AUDIO_AUDIO_MIXER_H_ 12 #define API_AUDIO_AUDIO_MIXER_H_ 13 14 #include <cstddef> 15 16 #include "api/audio/audio_frame.h" 17 #include "api/ref_count.h" 18 19 namespace webrtc { 20 21 // WORK IN PROGRESS 22 // This class is under development and is not yet intended for for use outside 23 // of WebRtc/Libjingle. 24 class AudioMixer : public RefCountInterface { 25 public: 26 // A callback class that all mixer participants must inherit from/implement. 27 class Source { 28 public: 29 enum class AudioFrameInfo { 30 kNormal, // The samples in audio_frame are valid and should be used. 31 kMuted, // The samples in audio_frame should not be used, but 32 // should be implicitly interpreted as zero. Other 33 // fields in audio_frame may be read and should 34 // contain meaningful values. 35 kError, // The audio_frame will not be used. 36 }; 37 38 // Overwrites `audio_frame`. The data_ field is overwritten with 39 // 10 ms of new audio (either 1 or 2 interleaved channels) at 40 // `sample_rate_hz`. All fields in `audio_frame` must be updated. 41 virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, 42 AudioFrame* audio_frame) = 0; 43 44 // A way for a mixer implementation to distinguish participants. 45 virtual int Ssrc() const = 0; 46 47 // A way for this source to say that GetAudioFrameWithInfo called 48 // with this sample rate or higher will not cause quality loss. 49 virtual int PreferredSampleRate() const = 0; 50 51 virtual ~Source() {} 52 }; 53 54 // Returns true if adding was successful. A source is never added 55 // twice. Addition and removal can happen on different threads. 56 virtual bool AddSource(Source* audio_source) = 0; 57 58 // Removal is never attempted if a source has not been successfully 59 // added to the mixer. 60 virtual void RemoveSource(Source* audio_source) = 0; 61 62 // Performs mixing by asking registered audio sources for audio. The 63 // mixed result is placed in the provided AudioFrame. This method 64 // will only be called from a single thread. The channels argument 65 // specifies the number of channels of the mix result. The mixer 66 // should mix at a rate that doesn't cause quality loss of the 67 // sources' audio. The mixing rate is one of the rates listed in 68 // AudioProcessing::NativeRate. All fields in 69 // `audio_frame_for_mixing` must be updated. 70 virtual void Mix(size_t number_of_channels, 71 AudioFrame* audio_frame_for_mixing) = 0; 72 73 protected: 74 // Since the mixer is reference counted, the destructor may be 75 // called from any thread. 76 ~AudioMixer() override {} 77 }; 78 } // namespace webrtc 79 80 #endif // API_AUDIO_AUDIO_MIXER_H_