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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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 #ifndef WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ | 11 #ifndef WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ |
12 #define WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ | 12 #define WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ |
13 | 13 |
14 #include "webrtc/audio_receive_stream.h" | 14 #include "webrtc/audio_receive_stream.h" |
15 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" | 15 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
16 | 16 |
17 namespace webrtc { | 17 namespace webrtc { |
18 | 18 |
19 class RemoteBitrateEstimator; | 19 class RemoteBitrateEstimator; |
20 | 20 |
21 namespace internal { | 21 namespace internal { |
22 | 22 |
23 class AudioReceiveStream : public webrtc::AudioReceiveStream { | 23 class AudioReceiveStream : public webrtc::AudioReceiveStream { |
24 public: | 24 public: |
25 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, | 25 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, |
26 const webrtc::AudioReceiveStream::Config& config); | 26 const webrtc::AudioReceiveStream::Config& config); |
27 ~AudioReceiveStream() override {} | 27 ~AudioReceiveStream() override {} |
28 | 28 |
| 29 // webrtc::ReceiveStream implementation. |
| 30 void Start() override; |
| 31 void Stop() override; |
| 32 void SignalNetworkState(NetworkState state) override; |
| 33 bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
| 34 bool DeliverRtp(const uint8_t* packet, size_t length) override; |
| 35 |
| 36 // webrtc::AudioReceiveStream implementation. |
29 webrtc::AudioReceiveStream::Stats GetStats() const override; | 37 webrtc::AudioReceiveStream::Stats GetStats() const override; |
30 | 38 |
31 bool DeliverRtcp(const uint8_t* packet, size_t length); | |
32 bool DeliverRtp(const uint8_t* packet, size_t length); | |
33 | |
34 const webrtc::AudioReceiveStream::Config& config() const { | 39 const webrtc::AudioReceiveStream::Config& config() const { |
35 return config_; | 40 return config_; |
36 } | 41 } |
37 | 42 |
38 private: | 43 private: |
39 RemoteBitrateEstimator* const remote_bitrate_estimator_; | 44 RemoteBitrateEstimator* const remote_bitrate_estimator_; |
40 const webrtc::AudioReceiveStream::Config config_; | 45 const webrtc::AudioReceiveStream::Config config_; |
41 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; | 46 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
42 }; | 47 }; |
43 } // namespace internal | 48 } // namespace internal |
44 } // namespace webrtc | 49 } // namespace webrtc |
45 | 50 |
46 #endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ | 51 #endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ |
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