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 AUDIO_AUDIO_RECEIVE_STREAM_H_
12 #define AUDIO_AUDIO_RECEIVE_STREAM_H_
13 
14 #include <memory>
15 #include <vector>
16 
17 #include "api/audio/audio_mixer.h"
18 #include "audio/audio_state.h"
19 #include "call/audio_receive_stream.h"
20 #include "call/rtp_packet_sink_interface.h"
21 #include "call/syncable.h"
22 #include "rtc_base/constructormagic.h"
23 #include "rtc_base/thread_checker.h"
24 
25 namespace webrtc {
26 class PacketRouter;
27 class RtcEventLog;
28 class RtpPacketReceived;
29 class RtpStreamReceiverControllerInterface;
30 class RtpStreamReceiverInterface;
31 
32 namespace voe {
33 class ChannelProxy;
34 }  // namespace voe
35 
36 namespace internal {
37 class AudioSendStream;
38 
39 class AudioReceiveStream final : public webrtc::AudioReceiveStream,
40                                  public AudioMixer::Source,
41                                  public Syncable {
42  public:
43   AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
44                      PacketRouter* packet_router,
45                      const webrtc::AudioReceiveStream::Config& config,
46                      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
47                      webrtc::RtcEventLog* event_log);
48   ~AudioReceiveStream() override;
49 
50   // webrtc::AudioReceiveStream implementation.
51   void Start() override;
52   void Stop() override;
53   webrtc::AudioReceiveStream::Stats GetStats() const override;
54   int GetOutputLevel() const override;
55   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
56   void SetGain(float gain) override;
57   std::vector<webrtc::RtpSource> GetSources() const override;
58 
59   // TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
60   // method shouldn't be needed. But it's currently used by the
61   // AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
62   // shuld be refactored or deleted, and then delete this method.
63   void OnRtpPacket(const RtpPacketReceived& packet);
64 
65   // AudioMixer::Source
66   AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
67                                        AudioFrame* audio_frame) override;
68   int Ssrc() const override;
69   int PreferredSampleRate() const override;
70 
71   // Syncable
72   int id() const override;
73   rtc::Optional<Syncable::Info> GetInfo() const override;
74   uint32_t GetPlayoutTimestamp() const override;
75   void SetMinimumPlayoutDelay(int delay_ms) override;
76 
77   void AssociateSendStream(AudioSendStream* send_stream);
78   void SignalNetworkState(NetworkState state);
79   bool DeliverRtcp(const uint8_t* packet, size_t length);
80   const webrtc::AudioReceiveStream::Config& config() const;
81 
82  private:
83   VoiceEngine* voice_engine() const;
84   AudioState* audio_state() const;
85   int SetVoiceEnginePlayout(bool playout);
86 
87   rtc::ThreadChecker worker_thread_checker_;
88   rtc::ThreadChecker module_process_thread_checker_;
89   const webrtc::AudioReceiveStream::Config config_;
90   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
91   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
92 
93   bool playing_ RTC_ACCESS_ON(worker_thread_checker_) = false;
94 
95   std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
96 
97   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
98 };
99 }  // namespace internal
100 }  // namespace webrtc
101 
102 #endif  // AUDIO_AUDIO_RECEIVE_STREAM_H_
103