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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_VOICE_ENGINE_CHANNEL_PROXY_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ |
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ |
13 | 13 |
14 #include "webrtc/api/audio/audio_mixer.h" | 14 #include "webrtc/api/audio/audio_mixer.h" |
15 #include "webrtc/base/constructormagic.h" | 15 #include "webrtc/base/constructormagic.h" |
16 #include "webrtc/base/race_checker.h" | 16 #include "webrtc/base/race_checker.h" |
17 #include "webrtc/base/thread_checker.h" | 17 #include "webrtc/base/thread_checker.h" |
18 #include "webrtc/call/rtp_transport_controller_receive.h" | |
18 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
19 #include "webrtc/voice_engine/channel_manager.h" | 20 #include "webrtc/voice_engine/channel_manager.h" |
20 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 21 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
21 | 22 |
22 #include <memory> | 23 #include <memory> |
23 #include <string> | 24 #include <string> |
24 #include <vector> | 25 #include <vector> |
25 | 26 |
26 namespace webrtc { | 27 namespace webrtc { |
27 | 28 |
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42 | 43 |
43 class Channel; | 44 class Channel; |
44 | 45 |
45 // This class provides the "view" of a voe::Channel that we need to implement | 46 // This class provides the "view" of a voe::Channel that we need to implement |
46 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two | 47 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two |
47 // purposes: | 48 // purposes: |
48 // 1. Allow mocking just the interfaces used, instead of the entire | 49 // 1. Allow mocking just the interfaces used, instead of the entire |
49 // voe::Channel class. | 50 // voe::Channel class. |
50 // 2. Provide a refined interface for the stream classes, including assumptions | 51 // 2. Provide a refined interface for the stream classes, including assumptions |
51 // on return values and input adaptation. | 52 // on return values and input adaptation. |
52 class ChannelProxy { | 53 class ChannelProxy : public RtpPacketSinkInterface { |
the sun
2017/04/18 09:59:00
No, I don't think you should register the ChannelP
nisse-webrtc
2017/04/19 08:35:43
I'll give it a try.
| |
53 public: | 54 public: |
54 ChannelProxy(); | 55 ChannelProxy(); |
55 explicit ChannelProxy(const ChannelOwner& channel_owner); | 56 explicit ChannelProxy(const ChannelOwner& channel_owner); |
56 virtual ~ChannelProxy(); | 57 virtual ~ChannelProxy(); |
57 | 58 |
58 virtual bool SetEncoder(int payload_type, | 59 virtual bool SetEncoder(int payload_type, |
59 std::unique_ptr<AudioEncoder> encoder); | 60 std::unique_ptr<AudioEncoder> encoder); |
60 | 61 |
61 virtual void SetRTCPStatus(bool enable); | 62 virtual void SetRTCPStatus(bool enable); |
62 virtual void SetLocalSSRC(uint32_t ssrc); | 63 virtual void SetLocalSSRC(uint32_t ssrc); |
63 virtual void SetRTCP_CNAME(const std::string& c_name); | 64 virtual void SetRTCP_CNAME(const std::string& c_name); |
64 virtual void SetNACKStatus(bool enable, int max_packets); | 65 virtual void SetNACKStatus(bool enable, int max_packets); |
65 virtual void SetSendAudioLevelIndicationStatus(bool enable, int id); | 66 virtual void SetSendAudioLevelIndicationStatus(bool enable, int id); |
66 virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id); | |
67 virtual void EnableSendTransportSequenceNumber(int id); | 67 virtual void EnableSendTransportSequenceNumber(int id); |
68 virtual void EnableReceiveTransportSequenceNumber(int id); | |
69 virtual void RegisterSenderCongestionControlObjects( | 68 virtual void RegisterSenderCongestionControlObjects( |
70 RtpTransportControllerSendInterface* transport, | 69 RtpTransportControllerSendInterface* transport, |
71 RtcpBandwidthObserver* bandwidth_observer); | 70 RtcpBandwidthObserver* bandwidth_observer); |
72 virtual void RegisterReceiverCongestionControlObjects( | 71 virtual void RegisterReceiverCongestionControlObjects( |
73 PacketRouter* packet_router); | 72 PacketRouter* packet_router); |
74 virtual void ResetSenderCongestionControlObjects(); | 73 virtual void ResetSenderCongestionControlObjects(); |
75 virtual void ResetReceiverCongestionControlObjects(); | 74 virtual void ResetReceiverCongestionControlObjects(); |
76 virtual CallStatistics GetRTCPStatistics() const; | 75 virtual CallStatistics GetRTCPStatistics() const; |
77 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; | 76 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; |
78 virtual NetworkStatistics GetNetworkStatistics() const; | 77 virtual NetworkStatistics GetNetworkStatistics() const; |
79 virtual AudioDecodingCallStats GetDecodingCallStatistics() const; | 78 virtual AudioDecodingCallStats GetDecodingCallStatistics() const; |
80 virtual int GetSpeechOutputLevel() const; | 79 virtual int GetSpeechOutputLevel() const; |
81 virtual int GetSpeechOutputLevelFullRange() const; | 80 virtual int GetSpeechOutputLevelFullRange() const; |
82 virtual uint32_t GetDelayEstimate() const; | 81 virtual uint32_t GetDelayEstimate() const; |
83 virtual bool SetSendTelephoneEventPayloadType(int payload_type, | 82 virtual bool SetSendTelephoneEventPayloadType(int payload_type, |
84 int payload_frequency); | 83 int payload_frequency); |
85 virtual bool SendTelephoneEventOutband(int event, int duration_ms); | 84 virtual bool SendTelephoneEventOutband(int event, int duration_ms); |
86 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms); | 85 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms); |
87 virtual void SetRecPayloadType(int payload_type, | 86 virtual void SetRecPayloadType(int payload_type, |
88 const SdpAudioFormat& format); | 87 const SdpAudioFormat& format); |
89 virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs); | 88 virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs); |
90 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink); | 89 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink); |
91 virtual void SetInputMute(bool muted); | 90 virtual void SetInputMute(bool muted); |
92 virtual void RegisterExternalTransport(Transport* transport); | 91 virtual void RegisterExternalTransport(Transport* transport); |
93 virtual void DeRegisterExternalTransport(); | 92 virtual void DeRegisterExternalTransport(); |
94 virtual void OnRtpPacket(const RtpPacketReceived& packet); | 93 // RtpPacketSinkInterface implementation. |
94 void OnRtpPacket(const RtpPacketReceived& packet) override; | |
95 virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length); | 95 virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length); |
96 virtual const rtc::scoped_refptr<AudioDecoderFactory>& | 96 virtual const rtc::scoped_refptr<AudioDecoderFactory>& |
97 GetAudioDecoderFactory() const; | 97 GetAudioDecoderFactory() const; |
98 virtual void SetChannelOutputVolumeScaling(float scaling); | 98 virtual void SetChannelOutputVolumeScaling(float scaling); |
99 virtual void SetRtcEventLog(RtcEventLog* event_log); | 99 virtual void SetRtcEventLog(RtcEventLog* event_log); |
100 virtual void EnableAudioNetworkAdaptor(const std::string& config_string); | 100 virtual void EnableAudioNetworkAdaptor(const std::string& config_string); |
101 virtual void DisableAudioNetworkAdaptor(); | 101 virtual void DisableAudioNetworkAdaptor(); |
102 virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, | 102 virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, |
103 int max_frame_length_ms); | 103 int max_frame_length_ms); |
104 virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( | 104 virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
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142 rtc::RaceChecker audio_thread_race_checker_; | 142 rtc::RaceChecker audio_thread_race_checker_; |
143 rtc::RaceChecker video_capture_thread_race_checker_; | 143 rtc::RaceChecker video_capture_thread_race_checker_; |
144 ChannelOwner channel_owner_; | 144 ChannelOwner channel_owner_; |
145 | 145 |
146 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy); | 146 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy); |
147 }; | 147 }; |
148 } // namespace voe | 148 } // namespace voe |
149 } // namespace webrtc | 149 } // namespace webrtc |
150 | 150 |
151 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ | 151 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ |
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