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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_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/audio_receive_stream.h" | 16 #include "webrtc/audio_receive_stream.h" |
17 #include "webrtc/audio_state.h" | 17 #include "webrtc/audio_state.h" |
18 #include "webrtc/base/thread_checker.h" | 18 #include "webrtc/base/thread_checker.h" |
19 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 19 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
20 | 20 |
21 namespace webrtc { | 21 namespace webrtc { |
22 class CongestionController; | 22 class CongestionController; |
23 class RemoteBitrateEstimator; | 23 class RemoteBitrateEstimator; |
24 class RtcEventLog; | |
24 | 25 |
25 namespace voe { | 26 namespace voe { |
26 class ChannelProxy; | 27 class ChannelProxy; |
27 } // namespace voe | 28 } // namespace voe |
28 | 29 |
29 namespace internal { | 30 namespace internal { |
30 | 31 |
31 class AudioReceiveStream final : public webrtc::AudioReceiveStream { | 32 class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
32 public: | 33 public: |
33 AudioReceiveStream(CongestionController* congestion_controller, | 34 AudioReceiveStream(CongestionController* congestion_controller, |
(...skipping 10 matching lines...) Expand all Loading... | |
44 size_t length, | 45 size_t length, |
45 const PacketTime& packet_time) override; | 46 const PacketTime& packet_time) override; |
46 | 47 |
47 // webrtc::AudioReceiveStream implementation. | 48 // webrtc::AudioReceiveStream implementation. |
48 webrtc::AudioReceiveStream::Stats GetStats() const override; | 49 webrtc::AudioReceiveStream::Stats GetStats() const override; |
49 | 50 |
50 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; | 51 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
51 | 52 |
52 const webrtc::AudioReceiveStream::Config& config() const; | 53 const webrtc::AudioReceiveStream::Config& config() const; |
53 | 54 |
55 void SetRtcEventLog(webrtc::RtcEventLog* event_log); | |
the sun
2016/03/03 09:25:12
No need for this - pass it in the ctor instead.
ivoc
2016/03/10 13:15:36
Good point.
| |
56 | |
54 private: | 57 private: |
55 VoiceEngine* voice_engine() const; | 58 VoiceEngine* voice_engine() const; |
56 | 59 |
57 rtc::ThreadChecker thread_checker_; | 60 rtc::ThreadChecker thread_checker_; |
58 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; | 61 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; |
59 const webrtc::AudioReceiveStream::Config config_; | 62 const webrtc::AudioReceiveStream::Config config_; |
60 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 63 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
61 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; | 64 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
62 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 65 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
63 | 66 |
64 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 67 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
65 }; | 68 }; |
66 } // namespace internal | 69 } // namespace internal |
67 } // namespace webrtc | 70 } // namespace webrtc |
68 | 71 |
69 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 72 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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