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1 /* | 1 /* |
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 // This file contains classes that implement RtpSenderInterface. | 11 // This file contains classes that implement RtpSenderInterface. |
12 // An RtpSender associates a MediaStreamTrackInterface with an underlying | 12 // An RtpSender associates a MediaStreamTrackInterface with an underlying |
13 // transport (provided by AudioProviderInterface/VideoProviderInterface) | 13 // transport (provided by AudioProviderInterface/VideoProviderInterface) |
14 | 14 |
15 #ifndef WEBRTC_API_RTPSENDER_H_ | 15 #ifndef WEBRTC_API_RTPSENDER_H_ |
16 #define WEBRTC_API_RTPSENDER_H_ | 16 #define WEBRTC_API_RTPSENDER_H_ |
17 | 17 |
18 #include <string> | 18 #include <string> |
19 | 19 |
20 #include "webrtc/api/mediastreamprovider.h" | 20 #include "webrtc/api/mediastreamprovider.h" |
21 #include "webrtc/api/rtpsenderinterface.h" | 21 #include "webrtc/api/rtpsenderinterface.h" |
22 #include "webrtc/api/statscollector.h" | 22 #include "webrtc/api/statscollector.h" |
23 #include "webrtc/base/basictypes.h" | 23 #include "webrtc/base/basictypes.h" |
24 #include "webrtc/base/criticalsection.h" | 24 #include "webrtc/base/criticalsection.h" |
25 #include "webrtc/base/scoped_ptr.h" | 25 #include "webrtc/base/scoped_ptr.h" |
26 #include "webrtc/media/base/audiorenderer.h" | 26 #include "webrtc/media/base/audiosource.h" |
27 | 27 |
28 namespace webrtc { | 28 namespace webrtc { |
29 | 29 |
30 // LocalAudioSinkAdapter receives data callback as a sink to the local | 30 // LocalAudioSinkAdapter receives data callback as a sink to the local |
31 // AudioTrack, and passes the data to the sink of AudioRenderer. | 31 // AudioTrack, and passes the data to the sink of AudioSource. |
32 class LocalAudioSinkAdapter : public AudioTrackSinkInterface, | 32 class LocalAudioSinkAdapter : public AudioTrackSinkInterface, |
33 public cricket::AudioRenderer { | 33 public cricket::AudioSource { |
34 public: | 34 public: |
35 LocalAudioSinkAdapter(); | 35 LocalAudioSinkAdapter(); |
36 virtual ~LocalAudioSinkAdapter(); | 36 virtual ~LocalAudioSinkAdapter(); |
37 | 37 |
38 private: | 38 private: |
39 // AudioSinkInterface implementation. | 39 // AudioSinkInterface implementation. |
40 void OnData(const void* audio_data, | 40 void OnData(const void* audio_data, |
41 int bits_per_sample, | 41 int bits_per_sample, |
42 int sample_rate, | 42 int sample_rate, |
43 size_t number_of_channels, | 43 size_t number_of_channels, |
44 size_t number_of_frames) override; | 44 size_t number_of_frames) override; |
45 | 45 |
46 // cricket::AudioRenderer implementation. | 46 // cricket::AudioSource implementation. |
47 void SetSink(cricket::AudioRenderer::Sink* sink) override; | 47 void SetSink(cricket::AudioSource::Sink* sink) override; |
48 | 48 |
49 cricket::AudioRenderer::Sink* sink_; | 49 cricket::AudioSource::Sink* sink_; |
50 // Critical section protecting |sink_|. | 50 // Critical section protecting |sink_|. |
51 rtc::CriticalSection lock_; | 51 rtc::CriticalSection lock_; |
52 }; | 52 }; |
53 | 53 |
54 class AudioRtpSender : public ObserverInterface, | 54 class AudioRtpSender : public ObserverInterface, |
55 public rtc::RefCountedObject<RtpSenderInterface> { | 55 public rtc::RefCountedObject<RtpSenderInterface> { |
56 public: | 56 public: |
57 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called | 57 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called |
58 // at the appropriate times. | 58 // at the appropriate times. |
59 AudioRtpSender(AudioTrackInterface* track, | 59 AudioRtpSender(AudioTrackInterface* track, |
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106 std::string id_; | 106 std::string id_; |
107 std::string stream_id_; | 107 std::string stream_id_; |
108 AudioProviderInterface* provider_; | 108 AudioProviderInterface* provider_; |
109 StatsCollector* stats_; | 109 StatsCollector* stats_; |
110 rtc::scoped_refptr<AudioTrackInterface> track_; | 110 rtc::scoped_refptr<AudioTrackInterface> track_; |
111 uint32_t ssrc_ = 0; | 111 uint32_t ssrc_ = 0; |
112 bool cached_track_enabled_ = false; | 112 bool cached_track_enabled_ = false; |
113 bool stopped_ = false; | 113 bool stopped_ = false; |
114 | 114 |
115 // Used to pass the data callback from the |track_| to the other end of | 115 // Used to pass the data callback from the |track_| to the other end of |
116 // cricket::AudioRenderer. | 116 // cricket::AudioSource. |
117 rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_; | 117 rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_; |
118 }; | 118 }; |
119 | 119 |
120 class VideoRtpSender : public ObserverInterface, | 120 class VideoRtpSender : public ObserverInterface, |
121 public rtc::RefCountedObject<RtpSenderInterface> { | 121 public rtc::RefCountedObject<RtpSenderInterface> { |
122 public: | 122 public: |
123 VideoRtpSender(VideoTrackInterface* track, | 123 VideoRtpSender(VideoTrackInterface* track, |
124 const std::string& stream_id, | 124 const std::string& stream_id, |
125 VideoProviderInterface* provider); | 125 VideoProviderInterface* provider); |
126 | 126 |
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169 VideoProviderInterface* provider_; | 169 VideoProviderInterface* provider_; |
170 rtc::scoped_refptr<VideoTrackInterface> track_; | 170 rtc::scoped_refptr<VideoTrackInterface> track_; |
171 uint32_t ssrc_ = 0; | 171 uint32_t ssrc_ = 0; |
172 bool cached_track_enabled_ = false; | 172 bool cached_track_enabled_ = false; |
173 bool stopped_ = false; | 173 bool stopped_ = false; |
174 }; | 174 }; |
175 | 175 |
176 } // namespace webrtc | 176 } // namespace webrtc |
177 | 177 |
178 #endif // WEBRTC_API_RTPSENDER_H_ | 178 #endif // WEBRTC_API_RTPSENDER_H_ |
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