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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 WEBRTC_API_AUDIO_AUDIO_MIXER_H_
12 : #define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
13 :
14 : #include <memory>
15 :
16 : #include "webrtc/base/refcount.h"
17 : #include "webrtc/modules/include/module_common_types.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 0 : class AudioMixer : public rtc::RefCountInterface {
25 : public:
26 : // A callback class that all mixer participants must inherit from/implement.
27 0 : 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 0 : 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 0 : ~AudioMixer() override {}
77 : };
78 : } // namespace webrtc
79 :
80 : #endif // WEBRTC_API_AUDIO_AUDIO_MIXER_H_
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