talk/app/webrtc/test/peerconnectiontestwrapper.h - Issue 1236023010: In PeerConnectionTestWrapper, put audio input on a separate thread.

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Side by Side Diff: talk/app/webrtc/test/peerconnectiontestwrapper.h

Issue 1236023010: In PeerConnectionTestWrapper, put audio input on a separate thread. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Fixing potential deadlock in FakeAudioCaptureModule, caused by unnecessary call to Thread::Send Created 5 years, 5 months ago

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1 /*	1 /*

2 * libjingle	2 * libjingle

3 * Copyright 2013 Google Inc.	3 * Copyright 2013 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,

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103 void SetLocalDescription(const std::string& type, const std::string& sdp);	103 void SetLocalDescription(const std::string& type, const std::string& sdp);

104 void SetRemoteDescription(const std::string& type, const std::string& sdp);	104 void SetRemoteDescription(const std::string& type, const std::string& sdp);

105 bool CheckForConnection();	105 bool CheckForConnection();

106 bool CheckForAudio();	106 bool CheckForAudio();

107 bool CheckForVideo();	107 bool CheckForVideo();

108 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(	108 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(

109 bool audio, const webrtc::FakeConstraints& audio_constraints,	109 bool audio, const webrtc::FakeConstraints& audio_constraints,

110 bool video, const webrtc::FakeConstraints& video_constraints);	110 bool video, const webrtc::FakeConstraints& video_constraints);

111	111

112 std::string name_;	112 std::string name_;

	113 rtc::scoped_ptr<rtc::Thread> media_input_thread_;

113 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>	114 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>

114 allocator_factory_;	115 allocator_factory_;

115 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;	116 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;

116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>	117 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>

117 peer_connection_factory_;	118 peer_connection_factory_;

118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;	119 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;

119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;	120 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;

120 };	121 };

121	122

122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_	123 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_

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