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Side by Side Diff: webrtc/api/peerconnectionfactory.h

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: More fixes for bots. Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2011 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
(...skipping 65 matching lines...) Expand 10 before | Expand all | Expand 10 after
76 rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack( 76 rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
77 const std::string& id, 77 const std::string& id,
78 VideoTrackSourceInterface* video_source) override; 78 VideoTrackSourceInterface* video_source) override;
79 79
80 rtc::scoped_refptr<AudioTrackInterface> 80 rtc::scoped_refptr<AudioTrackInterface>
81 CreateAudioTrack(const std::string& id, 81 CreateAudioTrack(const std::string& id,
82 AudioSourceInterface* audio_source) override; 82 AudioSourceInterface* audio_source) override;
83 83
84 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override; 84 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
85 void StopAecDump() override; 85 void StopAecDump() override;
86 bool StartRtcEventLog(rtc::PlatformFile file) override; 86 // TODO(ivoc) Remove after Chrome is updated.
87 void StopRtcEventLog() override; 87 bool StartRtcEventLog(rtc::PlatformFile file) override { return false; }
88 // TODO(ivoc) Remove after Chrome is updated.
89 void StopRtcEventLog() override {}
88 90
89 virtual webrtc::MediaControllerInterface* CreateMediaController( 91 virtual webrtc::MediaControllerInterface* CreateMediaController(
90 const cricket::MediaConfig& config) const; 92 const cricket::MediaConfig& config) const;
91 virtual rtc::Thread* signaling_thread(); 93 virtual rtc::Thread* signaling_thread();
92 virtual rtc::Thread* worker_thread(); 94 virtual rtc::Thread* worker_thread();
93 const Options& options() const { return options_; } 95 const Options& options() const { return options_; }
94 96
95 protected: 97 protected:
96 PeerConnectionFactory(); 98 PeerConnectionFactory();
97 PeerConnectionFactory( 99 PeerConnectionFactory(
(...skipping 25 matching lines...) Expand all
123 video_decoder_factory_; 125 video_decoder_factory_;
124 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_; 126 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
125 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; 127 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
126 128
127 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; 129 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
128 }; 130 };
129 131
130 } // namespace webrtc 132 } // namespace webrtc
131 133
132 #endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_ 134 #endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_
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