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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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