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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 "webrtc/audio_receive_stream.h" | 14 #include "webrtc/audio_receive_stream.h" |
15 #include "webrtc/audio_state.h" | 15 #include "webrtc/audio_state.h" |
16 #include "webrtc/base/thread_checker.h" | 16 #include "webrtc/base/thread_checker.h" |
17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 | |
21 class RemoteBitrateEstimator; | 20 class RemoteBitrateEstimator; |
22 | 21 |
| 22 namespace voe { |
| 23 class ChannelProxy; |
| 24 } // namespace voe |
| 25 |
23 namespace internal { | 26 namespace internal { |
24 | |
25 class AudioReceiveStream final : public webrtc::AudioReceiveStream { | 27 class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
26 public: | 28 public: |
27 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, | 29 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, |
28 const webrtc::AudioReceiveStream::Config& config, | 30 const webrtc::AudioReceiveStream::Config& config, |
29 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); | 31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); |
30 ~AudioReceiveStream() override; | 32 ~AudioReceiveStream() override; |
31 | 33 |
32 // webrtc::ReceiveStream implementation. | 34 // webrtc::ReceiveStream implementation. |
33 void Start() override; | 35 void Start() override; |
34 void Stop() override; | 36 void Stop() override; |
35 void SignalNetworkState(NetworkState state) override; | 37 void SignalNetworkState(NetworkState state) override; |
36 bool DeliverRtcp(const uint8_t* packet, size_t length) override; | 38 bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
37 bool DeliverRtp(const uint8_t* packet, | 39 bool DeliverRtp(const uint8_t* packet, |
38 size_t length, | 40 size_t length, |
39 const PacketTime& packet_time) override; | 41 const PacketTime& packet_time) override; |
40 | 42 |
41 // webrtc::AudioReceiveStream implementation. | 43 // webrtc::AudioReceiveStream implementation. |
42 webrtc::AudioReceiveStream::Stats GetStats() const override; | 44 webrtc::AudioReceiveStream::Stats GetStats() const override; |
43 | 45 |
44 const webrtc::AudioReceiveStream::Config& config() const; | 46 const webrtc::AudioReceiveStream::Config& config() const; |
45 | 47 |
46 private: | 48 private: |
47 VoiceEngine* voice_engine() const; | 49 VoiceEngine* voice_engine() const; |
48 | 50 |
49 rtc::ThreadChecker thread_checker_; | 51 rtc::ThreadChecker thread_checker_; |
50 RemoteBitrateEstimator* const remote_bitrate_estimator_; | 52 RemoteBitrateEstimator* const remote_bitrate_estimator_; |
51 const webrtc::AudioReceiveStream::Config config_; | 53 const webrtc::AudioReceiveStream::Config config_; |
52 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 54 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
53 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; | 55 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
| 56 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_; |
54 | 57 |
55 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 58 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
56 }; | 59 }; |
57 } // namespace internal | 60 } // namespace internal |
58 } // namespace webrtc | 61 } // namespace webrtc |
59 | 62 |
60 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 63 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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