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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/scoped_voe_interface.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 #include "webrtc/voice_engine/include/voe_base.h" | |
19 | 18 |
20 namespace webrtc { | 19 namespace webrtc { |
21 | 20 |
22 class RemoteBitrateEstimator; | 21 class RemoteBitrateEstimator; |
23 class VoiceEngine; | |
24 | 22 |
25 namespace internal { | 23 namespace internal { |
26 | 24 |
27 class AudioReceiveStream final : public webrtc::AudioReceiveStream { | 25 class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
28 public: | 26 public: |
29 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, | 27 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, |
30 const webrtc::AudioReceiveStream::Config& config, | 28 const webrtc::AudioReceiveStream::Config& config, |
31 VoiceEngine* voice_engine); | 29 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); |
32 ~AudioReceiveStream() override; | 30 ~AudioReceiveStream() override; |
33 | 31 |
34 // webrtc::ReceiveStream implementation. | 32 // webrtc::ReceiveStream implementation. |
35 void Start() override; | 33 void Start() override; |
36 void Stop() override; | 34 void Stop() override; |
37 void SignalNetworkState(NetworkState state) override; | 35 void SignalNetworkState(NetworkState state) override; |
38 bool DeliverRtcp(const uint8_t* packet, size_t length) override; | 36 bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
39 bool DeliverRtp(const uint8_t* packet, | 37 bool DeliverRtp(const uint8_t* packet, |
40 size_t length, | 38 size_t length, |
41 const PacketTime& packet_time) override; | 39 const PacketTime& packet_time) override; |
42 | 40 |
43 // webrtc::AudioReceiveStream implementation. | 41 // webrtc::AudioReceiveStream implementation. |
44 webrtc::AudioReceiveStream::Stats GetStats() const override; | 42 webrtc::AudioReceiveStream::Stats GetStats() const override; |
45 | 43 |
46 const webrtc::AudioReceiveStream::Config& config() const; | 44 const webrtc::AudioReceiveStream::Config& config() const; |
47 | 45 |
48 private: | 46 private: |
49 rtc::ThreadChecker thread_checker_; | 47 rtc::ThreadChecker thread_checker_; |
50 RemoteBitrateEstimator* const remote_bitrate_estimator_; | 48 RemoteBitrateEstimator* const remote_bitrate_estimator_; |
51 const webrtc::AudioReceiveStream::Config config_; | 49 const webrtc::AudioReceiveStream::Config config_; |
52 VoiceEngine* voice_engine_; | 50 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
53 // We hold one interface pointer to the VoE to make sure it is kept alive. | |
54 ScopedVoEInterface<VoEBase> voe_base_; | |
55 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; | 51 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
56 | 52 |
57 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 53 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
58 }; | 54 }; |
59 } // namespace internal | 55 } // namespace internal |
60 } // namespace webrtc | 56 } // namespace webrtc |
61 | 57 |
62 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 58 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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