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1 /*
2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3  *
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
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
13 
14 #include "webrtc/audio_receive_stream.h"
15 #include "webrtc/audio_state.h"
16 #include "webrtc/base/thread_checker.h"
17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
18 
19 namespace webrtc {
20 class CongestionController;
21 class RemoteBitrateEstimator;
22 
23 namespace voe {
24 class ChannelProxy;
25 }  // namespace voe
26 
27 namespace internal {
28 
29 class AudioReceiveStream final : public webrtc::AudioReceiveStream {
30  public:
31   AudioReceiveStream(CongestionController* congestion_controller,
32                      const webrtc::AudioReceiveStream::Config& config,
33                      const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
34   ~AudioReceiveStream() override;
35 
36   // webrtc::ReceiveStream implementation.
37   void Start() override;
38   void Stop() override;
39   void SignalNetworkState(NetworkState state) override;
40   bool DeliverRtcp(const uint8_t* packet, size_t length) override;
41   bool DeliverRtp(const uint8_t* packet,
42                   size_t length,
43                   const PacketTime& packet_time) override;
44 
45   // webrtc::AudioReceiveStream implementation.
46   webrtc::AudioReceiveStream::Stats GetStats() const override;
47 
48   void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
49 
50   const webrtc::AudioReceiveStream::Config& config() const;
51 
52  private:
53   VoiceEngine* voice_engine() const;
54 
55   rtc::ThreadChecker thread_checker_;
56   RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
57   const webrtc::AudioReceiveStream::Config config_;
58   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
59   rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
60   rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
61 
62   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
63 };
64 }  // namespace internal
65 }  // namespace webrtc
66 
67 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
68