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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 <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/api/audio/audio_mixer.h" | 16 #include "webrtc/api/audio/audio_mixer.h" |
17 #include "webrtc/audio/audio_state.h" | 17 #include "webrtc/audio/audio_state.h" |
18 #include "webrtc/base/constructormagic.h" | 18 #include "webrtc/base/constructormagic.h" |
19 #include "webrtc/base/thread_checker.h" | 19 #include "webrtc/base/thread_checker.h" |
20 #include "webrtc/call/audio_receive_stream.h" | 20 #include "webrtc/call/audio_receive_stream.h" |
21 #include "webrtc/call/rtp_transport_controller_receive.h" | |
21 #include "webrtc/call/syncable.h" | 22 #include "webrtc/call/syncable.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" | |
24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | |
the sun
2017/04/18 09:59:00
We only need forward decl here.
nisse-webrtc
2017/04/19 08:35:43
Done.
| |
22 | 25 |
23 namespace webrtc { | 26 namespace webrtc { |
24 class PacketRouter; | 27 class PacketRouter; |
25 class RtcEventLog; | 28 class RtcEventLog; |
26 class RtpPacketReceived; | 29 class RtpPacketReceived; |
27 | 30 |
28 namespace voe { | 31 namespace voe { |
29 class ChannelProxy; | 32 class ChannelProxy; |
30 } // namespace voe | 33 } // namespace voe |
31 | 34 |
32 namespace internal { | 35 namespace internal { |
33 class AudioSendStream; | 36 class AudioSendStream; |
34 | 37 |
35 class AudioReceiveStream final : public webrtc::AudioReceiveStream, | 38 class AudioReceiveStream final : public webrtc::RtpPacketReceiverInterface, |
39 public webrtc::AudioReceiveStream, | |
36 public AudioMixer::Source, | 40 public AudioMixer::Source, |
37 public Syncable { | 41 public Syncable { |
38 public: | 42 public: |
39 AudioReceiveStream(PacketRouter* packet_router, | 43 AudioReceiveStream(PacketRouter* packet_router, |
40 const webrtc::AudioReceiveStream::Config& config, | 44 const webrtc::AudioReceiveStream::Config& config, |
41 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 45 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
42 webrtc::RtcEventLog* event_log); | 46 webrtc::RtcEventLog* event_log); |
43 ~AudioReceiveStream() override; | 47 ~AudioReceiveStream() override; |
44 | 48 |
45 // webrtc::AudioReceiveStream implementation. | 49 // webrtc::AudioReceiveStream implementation. |
46 void Start() override; | 50 void Start() override; |
47 void Stop() override; | 51 void Stop() override; |
48 webrtc::AudioReceiveStream::Stats GetStats() const override; | 52 webrtc::AudioReceiveStream::Stats GetStats() const override; |
49 int GetOutputLevel() const override; | 53 int GetOutputLevel() const override; |
50 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; | 54 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
51 void SetGain(float gain) override; | 55 void SetGain(float gain) override; |
52 | 56 |
53 // TODO(nisse): Intended to be part of an RtpPacketReceiver interface. | 57 // webrtc::RtpPacketReceiverInterface implementation |
54 void OnRtpPacket(const RtpPacketReceived& packet); | 58 bool OnRtpPacketReceive(RtpPacketReceived* packet) override; |
55 | 59 |
56 // AudioMixer::Source | 60 // AudioMixer::Source |
57 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, | 61 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
58 AudioFrame* audio_frame) override; | 62 AudioFrame* audio_frame) override; |
59 int Ssrc() const override; | 63 int Ssrc() const override; |
60 int PreferredSampleRate() const override; | 64 int PreferredSampleRate() const override; |
61 | 65 |
62 // Syncable | 66 // Syncable |
63 int id() const override; | 67 int id() const override; |
64 rtc::Optional<Syncable::Info> GetInfo() const override; | 68 rtc::Optional<Syncable::Info> GetInfo() const override; |
65 uint32_t GetPlayoutTimestamp() const override; | 69 uint32_t GetPlayoutTimestamp() const override; |
66 void SetMinimumPlayoutDelay(int delay_ms) override; | 70 void SetMinimumPlayoutDelay(int delay_ms) override; |
67 | 71 |
68 void AssociateSendStream(AudioSendStream* send_stream); | 72 void AssociateSendStream(AudioSendStream* send_stream); |
69 void SignalNetworkState(NetworkState state); | 73 void SignalNetworkState(NetworkState state); |
70 bool DeliverRtcp(const uint8_t* packet, size_t length); | 74 bool DeliverRtcp(const uint8_t* packet, size_t length); |
71 const webrtc::AudioReceiveStream::Config& config() const; | 75 const webrtc::AudioReceiveStream::Config& config() const; |
72 | 76 |
73 private: | 77 private: |
74 VoiceEngine* voice_engine() const; | 78 VoiceEngine* voice_engine() const; |
75 AudioState* audio_state() const; | 79 AudioState* audio_state() const; |
76 int SetVoiceEnginePlayout(bool playout); | 80 int SetVoiceEnginePlayout(bool playout); |
77 | 81 |
78 rtc::ThreadChecker worker_thread_checker_; | 82 rtc::ThreadChecker worker_thread_checker_; |
79 rtc::ThreadChecker module_process_thread_checker_; | 83 rtc::ThreadChecker module_process_thread_checker_; |
80 const webrtc::AudioReceiveStream::Config config_; | 84 const webrtc::AudioReceiveStream::Config config_; |
85 RtpHeaderExtensionMap rtp_header_extensions_; | |
86 | |
81 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 87 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
82 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 88 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
83 | 89 |
84 bool playing_ ACCESS_ON(worker_thread_checker_) = false; | 90 bool playing_ ACCESS_ON(worker_thread_checker_) = false; |
85 | 91 |
86 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 92 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
87 }; | 93 }; |
88 } // namespace internal | 94 } // namespace internal |
89 } // namespace webrtc | 95 } // namespace webrtc |
90 | 96 |
91 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 97 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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