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Side by Side Diff: webrtc/call.h

Issue 2060403002: Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Revert fix for asan, protect instead. Added destruction observer to frames in ViEEncoder tests. Created 4 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #ifndef WEBRTC_CALL_H_ 10 #ifndef WEBRTC_CALL_H_
(...skipping 92 matching lines...) Expand 10 before | Expand all | Expand 10 after
103 virtual AudioSendStream* CreateAudioSendStream( 103 virtual AudioSendStream* CreateAudioSendStream(
104 const AudioSendStream::Config& config) = 0; 104 const AudioSendStream::Config& config) = 0;
105 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; 105 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
106 106
107 virtual AudioReceiveStream* CreateAudioReceiveStream( 107 virtual AudioReceiveStream* CreateAudioReceiveStream(
108 const AudioReceiveStream::Config& config) = 0; 108 const AudioReceiveStream::Config& config) = 0;
109 virtual void DestroyAudioReceiveStream( 109 virtual void DestroyAudioReceiveStream(
110 AudioReceiveStream* receive_stream) = 0; 110 AudioReceiveStream* receive_stream) = 0;
111 111
112 virtual VideoSendStream* CreateVideoSendStream( 112 virtual VideoSendStream* CreateVideoSendStream(
113 const VideoSendStream::Config& config, 113 VideoSendStream::Config config,
114 const VideoEncoderConfig& encoder_config) = 0; 114 VideoEncoderConfig encoder_config) = 0;
115 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; 115 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
116 116
117 virtual VideoReceiveStream* CreateVideoReceiveStream( 117 virtual VideoReceiveStream* CreateVideoReceiveStream(
118 VideoReceiveStream::Config configuration) = 0; 118 VideoReceiveStream::Config configuration) = 0;
119 virtual void DestroyVideoReceiveStream( 119 virtual void DestroyVideoReceiveStream(
120 VideoReceiveStream* receive_stream) = 0; 120 VideoReceiveStream* receive_stream) = 0;
121 121
122 // All received RTP and RTCP packets for the call should be inserted to this 122 // All received RTP and RTCP packets for the call should be inserted to this
123 // PacketReceiver. The PacketReceiver pointer is valid as long as the 123 // PacketReceiver. The PacketReceiver pointer is valid as long as the
124 // Call instance exists. 124 // Call instance exists.
(...skipping 26 matching lines...) Expand all
151 virtual bool StartEventLog(rtc::PlatformFile log_file, 151 virtual bool StartEventLog(rtc::PlatformFile log_file,
152 int64_t max_size_bytes) = 0; 152 int64_t max_size_bytes) = 0;
153 virtual void StopEventLog() = 0; 153 virtual void StopEventLog() = 0;
154 154
155 virtual ~Call() {} 155 virtual ~Call() {}
156 }; 156 };
157 157
158 } // namespace webrtc 158 } // namespace webrtc
159 159
160 #endif // WEBRTC_CALL_H_ 160 #endif // WEBRTC_CALL_H_
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