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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 |
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23 namespace webrtc { | 23 namespace webrtc { |
24 class CongestionController; | 24 class CongestionController; |
25 class RemoteBitrateEstimator; | 25 class RemoteBitrateEstimator; |
26 class RtcEventLog; | 26 class RtcEventLog; |
27 | 27 |
28 namespace voe { | 28 namespace voe { |
29 class ChannelProxy; | 29 class ChannelProxy; |
30 } // namespace voe | 30 } // namespace voe |
31 | 31 |
32 namespace internal { | 32 namespace internal { |
| 33 class AudioSendStream; |
33 | 34 |
34 class AudioReceiveStream final : public webrtc::AudioReceiveStream, | 35 class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
35 public AudioMixer::Source { | 36 public AudioMixer::Source { |
36 public: | 37 public: |
37 AudioReceiveStream(CongestionController* congestion_controller, | 38 AudioReceiveStream(CongestionController* congestion_controller, |
38 const webrtc::AudioReceiveStream::Config& config, | 39 const webrtc::AudioReceiveStream::Config& config, |
39 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 40 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
40 webrtc::RtcEventLog* event_log); | 41 webrtc::RtcEventLog* event_log); |
41 ~AudioReceiveStream() override; | 42 ~AudioReceiveStream() override; |
42 | 43 |
43 // webrtc::AudioReceiveStream implementation. | 44 // webrtc::AudioReceiveStream implementation. |
44 void Start() override; | 45 void Start() override; |
45 void Stop() override; | 46 void Stop() override; |
46 webrtc::AudioReceiveStream::Stats GetStats() const override; | 47 webrtc::AudioReceiveStream::Stats GetStats() const override; |
47 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; | 48 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
48 void SetGain(float gain) override; | 49 void SetGain(float gain) override; |
49 | 50 |
| 51 void AssociateSendStream(AudioSendStream* send_stream); |
50 void SignalNetworkState(NetworkState state); | 52 void SignalNetworkState(NetworkState state); |
51 bool DeliverRtcp(const uint8_t* packet, size_t length); | 53 bool DeliverRtcp(const uint8_t* packet, size_t length); |
52 bool DeliverRtp(const uint8_t* packet, | 54 bool DeliverRtp(const uint8_t* packet, |
53 size_t length, | 55 size_t length, |
54 const PacketTime& packet_time); | 56 const PacketTime& packet_time); |
55 const webrtc::AudioReceiveStream::Config& config() const; | 57 const webrtc::AudioReceiveStream::Config& config() const; |
56 | 58 |
57 // AudioMixer::Source | 59 // AudioMixer::Source |
58 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, | 60 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
59 AudioFrame* audio_frame) override; | 61 AudioFrame* audio_frame) override; |
60 int PreferredSampleRate() const override; | 62 int PreferredSampleRate() const override; |
61 int Ssrc() const override; | 63 int Ssrc() const override; |
62 | 64 |
63 private: | 65 private: |
64 VoiceEngine* voice_engine() const; | 66 VoiceEngine* voice_engine() const; |
65 | 67 |
66 rtc::ThreadChecker thread_checker_; | 68 rtc::ThreadChecker thread_checker_; |
67 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; | 69 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; |
68 const webrtc::AudioReceiveStream::Config config_; | 70 const webrtc::AudioReceiveStream::Config config_; |
69 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 71 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
70 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; | 72 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
71 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 73 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
72 | 74 |
73 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 75 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
74 }; | 76 }; |
75 } // namespace internal | 77 } // namespace internal |
76 } // namespace webrtc | 78 } // namespace webrtc |
77 | 79 |
78 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 80 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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