Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(260)

Side by Side Diff: webrtc/voice_engine/channel_proxy.h

Issue 2709723003: Initial implementation of RtpTransportControllerReceive and related interfaces.
Patch Set: Adapt Call to use the new RtpTransportReceive class. Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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/voice_engine/channel_manager.h" 19 #include "webrtc/voice_engine/channel_manager.h"
19 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 20 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
20 21
21 #include <memory> 22 #include <memory>
22 #include <string> 23 #include <string>
23 #include <vector> 24 #include <vector>
24 25
25 namespace webrtc { 26 namespace webrtc {
26 27
27 class AudioSinkInterface; 28 class AudioSinkInterface;
(...skipping 12 matching lines...) Expand all
40 41
41 class Channel; 42 class Channel;
42 43
43 // This class provides the "view" of a voe::Channel that we need to implement 44 // This class provides the "view" of a voe::Channel that we need to implement
44 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two 45 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
45 // purposes: 46 // purposes:
46 // 1. Allow mocking just the interfaces used, instead of the entire 47 // 1. Allow mocking just the interfaces used, instead of the entire
47 // voe::Channel class. 48 // voe::Channel class.
48 // 2. Provide a refined interface for the stream classes, including assumptions 49 // 2. Provide a refined interface for the stream classes, including assumptions
49 // on return values and input adaptation. 50 // on return values and input adaptation.
50 class ChannelProxy { 51 class ChannelProxy : public RtpPacketSinkInterface {
51 public: 52 public:
52 ChannelProxy(); 53 ChannelProxy();
53 explicit ChannelProxy(const ChannelOwner& channel_owner); 54 explicit ChannelProxy(const ChannelOwner& channel_owner);
54 virtual ~ChannelProxy(); 55 virtual ~ChannelProxy();
55 56
56 virtual void SetRTCPStatus(bool enable); 57 virtual void SetRTCPStatus(bool enable);
57 virtual void SetLocalSSRC(uint32_t ssrc); 58 virtual void SetLocalSSRC(uint32_t ssrc);
58 virtual void SetRTCP_CNAME(const std::string& c_name); 59 virtual void SetRTCP_CNAME(const std::string& c_name);
59 virtual void SetNACKStatus(bool enable, int max_packets); 60 virtual void SetNACKStatus(bool enable, int max_packets);
60 virtual void SetSendAudioLevelIndicationStatus(bool enable, int id); 61 virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
61 virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
62 virtual void EnableSendTransportSequenceNumber(int id); 62 virtual void EnableSendTransportSequenceNumber(int id);
63 virtual void EnableReceiveTransportSequenceNumber(int id);
64 virtual void RegisterSenderCongestionControlObjects( 63 virtual void RegisterSenderCongestionControlObjects(
65 RtpPacketSender* rtp_packet_sender, 64 RtpPacketSender* rtp_packet_sender,
66 TransportFeedbackObserver* transport_feedback_observer, 65 TransportFeedbackObserver* transport_feedback_observer,
67 PacketRouter* packet_router, 66 PacketRouter* packet_router,
68 RtcpBandwidthObserver* bandwidth_observer); 67 RtcpBandwidthObserver* bandwidth_observer);
69 virtual void RegisterReceiverCongestionControlObjects( 68 virtual void RegisterReceiverCongestionControlObjects(
70 PacketRouter* packet_router); 69 PacketRouter* packet_router);
71 virtual void ResetCongestionControlObjects(); 70 virtual void ResetCongestionControlObjects();
72 virtual CallStatistics GetRTCPStatistics() const; 71 virtual CallStatistics GetRTCPStatistics() const;
73 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; 72 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
74 virtual NetworkStatistics GetNetworkStatistics() const; 73 virtual NetworkStatistics GetNetworkStatistics() const;
75 virtual AudioDecodingCallStats GetDecodingCallStatistics() const; 74 virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
76 virtual int GetSpeechOutputLevel() const; 75 virtual int GetSpeechOutputLevel() const;
77 virtual int GetSpeechOutputLevelFullRange() const; 76 virtual int GetSpeechOutputLevelFullRange() const;
78 virtual uint32_t GetDelayEstimate() const; 77 virtual uint32_t GetDelayEstimate() const;
79 virtual bool SetSendTelephoneEventPayloadType(int payload_type, 78 virtual bool SetSendTelephoneEventPayloadType(int payload_type,
80 int payload_frequency); 79 int payload_frequency);
81 virtual bool SendTelephoneEventOutband(int event, int duration_ms); 80 virtual bool SendTelephoneEventOutband(int event, int duration_ms);
82 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms); 81 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
83 virtual void SetRecPayloadType(int payload_type, 82 virtual void SetRecPayloadType(int payload_type,
84 const SdpAudioFormat& format); 83 const SdpAudioFormat& format);
85 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink); 84 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
86 virtual void SetInputMute(bool muted); 85 virtual void SetInputMute(bool muted);
87 virtual void RegisterExternalTransport(Transport* transport); 86 virtual void RegisterExternalTransport(Transport* transport);
88 virtual void DeRegisterExternalTransport(); 87 virtual void DeRegisterExternalTransport();
89 virtual void OnRtpPacket(const RtpPacketReceived& packet); 88 // RtpPacketSinkInterface implementation.
89 void OnRtpPacket(const RtpPacketReceived& packet) override;
90 virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length); 90 virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
91 virtual const rtc::scoped_refptr<AudioDecoderFactory>& 91 virtual const rtc::scoped_refptr<AudioDecoderFactory>&
92 GetAudioDecoderFactory() const; 92 GetAudioDecoderFactory() const;
93 virtual void SetChannelOutputVolumeScaling(float scaling); 93 virtual void SetChannelOutputVolumeScaling(float scaling);
94 virtual void SetRtcEventLog(RtcEventLog* event_log); 94 virtual void SetRtcEventLog(RtcEventLog* event_log);
95 virtual void EnableAudioNetworkAdaptor(const std::string& config_string); 95 virtual void EnableAudioNetworkAdaptor(const std::string& config_string);
96 virtual void DisableAudioNetworkAdaptor(); 96 virtual void DisableAudioNetworkAdaptor();
97 virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, 97 virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
98 int max_frame_length_ms); 98 int max_frame_length_ms);
99 virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( 99 virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
(...skipping 33 matching lines...) Expand 10 before | Expand all | Expand 10 after
133 rtc::RaceChecker audio_thread_race_checker_; 133 rtc::RaceChecker audio_thread_race_checker_;
134 rtc::RaceChecker video_capture_thread_race_checker_; 134 rtc::RaceChecker video_capture_thread_race_checker_;
135 ChannelOwner channel_owner_; 135 ChannelOwner channel_owner_;
136 136
137 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy); 137 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
138 }; 138 };
139 } // namespace voe 139 } // namespace voe
140 } // namespace webrtc 140 } // namespace webrtc
141 141
142 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ 142 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698