Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(4)

Side by Side Diff: talk/media/webrtc/fakewebrtccall.h

Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: that's all folks! (incl rebase), or is it? Created 5 years, 1 month 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
« no previous file with comments | « talk/media/base/mediaengine.h ('k') | talk/media/webrtc/webrtcvoiceengine.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2015 Google Inc. 3 * Copyright 2015 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 29 matching lines...) Expand all
40 #include <vector> 40 #include <vector>
41 41
42 #include "webrtc/call.h" 42 #include "webrtc/call.h"
43 #include "webrtc/audio_receive_stream.h" 43 #include "webrtc/audio_receive_stream.h"
44 #include "webrtc/audio_send_stream.h" 44 #include "webrtc/audio_send_stream.h"
45 #include "webrtc/video_frame.h" 45 #include "webrtc/video_frame.h"
46 #include "webrtc/video_receive_stream.h" 46 #include "webrtc/video_receive_stream.h"
47 #include "webrtc/video_send_stream.h" 47 #include "webrtc/video_send_stream.h"
48 48
49 namespace cricket { 49 namespace cricket {
50 50 class FakeAudioSendStream final : public webrtc::AudioSendStream {
51 class FakeAudioSendStream : public webrtc::AudioSendStream {
52 public: 51 public:
53 explicit FakeAudioSendStream( 52 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
54 const webrtc::AudioSendStream::Config& config);
55 53
56 const webrtc::AudioSendStream::Config& GetConfig() const; 54 const webrtc::AudioSendStream::Config& GetConfig() const;
57 void SetStats(const webrtc::AudioSendStream::Stats& stats); 55 void SetStats(const webrtc::AudioSendStream::Stats& stats);
58 56
59 private: 57 private:
60 // webrtc::SendStream implementation. 58 // webrtc::SendStream implementation.
61 void Start() override {} 59 void Start() override {}
62 void Stop() override {} 60 void Stop() override {}
63 void SignalNetworkState(webrtc::NetworkState state) override {} 61 void SignalNetworkState(webrtc::NetworkState state) override {}
64 bool DeliverRtcp(const uint8_t* packet, size_t length) override { 62 bool DeliverRtcp(const uint8_t* packet, size_t length) override {
65 return true; 63 return true;
66 } 64 }
67 65
68 // webrtc::AudioSendStream implementation. 66 // webrtc::AudioSendStream implementation.
69 webrtc::AudioSendStream::Stats GetStats() const override; 67 webrtc::AudioSendStream::Stats GetStats() const override;
70 68
71 webrtc::AudioSendStream::Config config_; 69 webrtc::AudioSendStream::Config config_;
72 webrtc::AudioSendStream::Stats stats_; 70 webrtc::AudioSendStream::Stats stats_;
73 }; 71 };
74 72
75 class FakeAudioReceiveStream : public webrtc::AudioReceiveStream { 73 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
76 public: 74 public:
77 explicit FakeAudioReceiveStream( 75 explicit FakeAudioReceiveStream(
78 const webrtc::AudioReceiveStream::Config& config); 76 const webrtc::AudioReceiveStream::Config& config);
79 77
80 const webrtc::AudioReceiveStream::Config& GetConfig() const; 78 const webrtc::AudioReceiveStream::Config& GetConfig() const;
81 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); 79 void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
82 int received_packets() const { return received_packets_; } 80 int received_packets() const { return received_packets_; }
83 void IncrementReceivedPackets(); 81 void IncrementReceivedPackets();
84 82
85 private: 83 private:
(...skipping 11 matching lines...) Expand all
97 } 95 }
98 96
99 // webrtc::AudioReceiveStream implementation. 97 // webrtc::AudioReceiveStream implementation.
100 webrtc::AudioReceiveStream::Stats GetStats() const override; 98 webrtc::AudioReceiveStream::Stats GetStats() const override;
101 99
102 webrtc::AudioReceiveStream::Config config_; 100 webrtc::AudioReceiveStream::Config config_;
103 webrtc::AudioReceiveStream::Stats stats_; 101 webrtc::AudioReceiveStream::Stats stats_;
104 int received_packets_; 102 int received_packets_;
105 }; 103 };
106 104
107 class FakeVideoSendStream : public webrtc::VideoSendStream, 105 class FakeVideoSendStream final : public webrtc::VideoSendStream,
108 public webrtc::VideoCaptureInput { 106 public webrtc::VideoCaptureInput {
109 public: 107 public:
110 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, 108 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
111 const webrtc::VideoEncoderConfig& encoder_config); 109 const webrtc::VideoEncoderConfig& encoder_config);
112 webrtc::VideoSendStream::Config GetConfig() const; 110 webrtc::VideoSendStream::Config GetConfig() const;
113 webrtc::VideoEncoderConfig GetEncoderConfig() const; 111 webrtc::VideoEncoderConfig GetEncoderConfig() const;
114 std::vector<webrtc::VideoStream> GetVideoStreams(); 112 std::vector<webrtc::VideoStream> GetVideoStreams();
115 113
116 bool IsSending() const; 114 bool IsSending() const;
117 bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const; 115 bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
118 bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const; 116 bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
(...skipping 27 matching lines...) Expand all
146 bool codec_settings_set_; 144 bool codec_settings_set_;
147 union VpxSettings { 145 union VpxSettings {
148 webrtc::VideoCodecVP8 vp8; 146 webrtc::VideoCodecVP8 vp8;
149 webrtc::VideoCodecVP9 vp9; 147 webrtc::VideoCodecVP9 vp9;
150 } vpx_settings_; 148 } vpx_settings_;
151 int num_swapped_frames_; 149 int num_swapped_frames_;
152 webrtc::VideoFrame last_frame_; 150 webrtc::VideoFrame last_frame_;
153 webrtc::VideoSendStream::Stats stats_; 151 webrtc::VideoSendStream::Stats stats_;
154 }; 152 };
155 153
156 class FakeVideoReceiveStream : public webrtc::VideoReceiveStream { 154 class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
157 public: 155 public:
158 explicit FakeVideoReceiveStream( 156 explicit FakeVideoReceiveStream(
159 const webrtc::VideoReceiveStream::Config& config); 157 const webrtc::VideoReceiveStream::Config& config);
160 158
161 webrtc::VideoReceiveStream::Config GetConfig(); 159 webrtc::VideoReceiveStream::Config GetConfig();
162 160
163 bool IsReceiving() const; 161 bool IsReceiving() const;
164 162
165 void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms); 163 void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms);
166 164
(...skipping 14 matching lines...) Expand all
181 } 179 }
182 180
183 // webrtc::VideoReceiveStream implementation. 181 // webrtc::VideoReceiveStream implementation.
184 webrtc::VideoReceiveStream::Stats GetStats() const override; 182 webrtc::VideoReceiveStream::Stats GetStats() const override;
185 183
186 webrtc::VideoReceiveStream::Config config_; 184 webrtc::VideoReceiveStream::Config config_;
187 bool receiving_; 185 bool receiving_;
188 webrtc::VideoReceiveStream::Stats stats_; 186 webrtc::VideoReceiveStream::Stats stats_;
189 }; 187 };
190 188
191 class FakeCall : public webrtc::Call, public webrtc::PacketReceiver { 189 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
192 public: 190 public:
193 explicit FakeCall(const webrtc::Call::Config& config); 191 explicit FakeCall(const webrtc::Call::Config& config);
194 ~FakeCall() override; 192 ~FakeCall() override;
195 193
196 webrtc::Call::Config GetConfig() const; 194 webrtc::Call::Config GetConfig() const;
197 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); 195 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
198 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); 196 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
199 197
200 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); 198 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
201 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); 199 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
(...skipping 47 matching lines...) Expand 10 before | Expand all | Expand 10 after
249 std::vector<FakeAudioSendStream*> audio_send_streams_; 247 std::vector<FakeAudioSendStream*> audio_send_streams_;
250 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 248 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
251 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 249 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
252 250
253 int num_created_send_streams_; 251 int num_created_send_streams_;
254 int num_created_receive_streams_; 252 int num_created_receive_streams_;
255 }; 253 };
256 254
257 } // namespace cricket 255 } // namespace cricket
258 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 256 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
OLDNEW
« no previous file with comments | « talk/media/base/mediaengine.h ('k') | talk/media/webrtc/webrtcvoiceengine.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698