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Side by Side Diff: talk/app/webrtc/peerconnectionfactory.h

Issue 1413483003: Added option to specify a maximum file size when recording an AEC dump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added function to avoid breaking Chromium. Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2011 Google Inc. 3 * Copyright 2011 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 75 matching lines...) Expand 10 before | Expand all | Expand 10 after
86 const MediaConstraintsInterface* constraints) override; 86 const MediaConstraintsInterface* constraints) override;
87 87
88 rtc::scoped_refptr<VideoTrackInterface> 88 rtc::scoped_refptr<VideoTrackInterface>
89 CreateVideoTrack(const std::string& id, 89 CreateVideoTrack(const std::string& id,
90 VideoSourceInterface* video_source) override; 90 VideoSourceInterface* video_source) override;
91 91
92 rtc::scoped_refptr<AudioTrackInterface> 92 rtc::scoped_refptr<AudioTrackInterface>
93 CreateAudioTrack(const std::string& id, 93 CreateAudioTrack(const std::string& id,
94 AudioSourceInterface* audio_source) override; 94 AudioSourceInterface* audio_source) override;
95 95
96 bool StartAecDump(rtc::PlatformFile file) override; 96 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
97 void StopAecDump() override; 97 void StopAecDump() override;
98 bool StartRtcEventLog(rtc::PlatformFile file) override; 98 bool StartRtcEventLog(rtc::PlatformFile file) override;
99 void StopRtcEventLog() override; 99 void StopRtcEventLog() override;
100 100
101 virtual webrtc::MediaControllerInterface* CreateMediaController() const; 101 virtual webrtc::MediaControllerInterface* CreateMediaController() const;
102 virtual rtc::Thread* signaling_thread(); 102 virtual rtc::Thread* signaling_thread();
103 virtual rtc::Thread* worker_thread(); 103 virtual rtc::Thread* worker_thread();
104 const Options& options() const { return options_; } 104 const Options& options() const { return options_; }
105 105
106 protected: 106 protected:
(...skipping 28 matching lines...) Expand all
135 video_decoder_factory_; 135 video_decoder_factory_;
136 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_; 136 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
137 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; 137 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
138 138
139 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; 139 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
140 }; 140 };
141 141
142 } // namespace webrtc 142 } // namespace webrtc
143 143
144 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ 144 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
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