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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: Fix audio thread check when adding audio to bitrateallocator. Created 4 years, 4 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 95 matching lines...) Expand 10 before | Expand all | Expand 10 after
106 virtual AudioSendStream* CreateAudioSendStream( 106 virtual AudioSendStream* CreateAudioSendStream(
107 const AudioSendStream::Config& config) = 0; 107 const AudioSendStream::Config& config) = 0;
108 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; 108 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
109 109
110 virtual AudioReceiveStream* CreateAudioReceiveStream( 110 virtual AudioReceiveStream* CreateAudioReceiveStream(
111 const AudioReceiveStream::Config& config) = 0; 111 const AudioReceiveStream::Config& config) = 0;
112 virtual void DestroyAudioReceiveStream( 112 virtual void DestroyAudioReceiveStream(
113 AudioReceiveStream* receive_stream) = 0; 113 AudioReceiveStream* receive_stream) = 0;
114 114
115 virtual VideoSendStream* CreateVideoSendStream( 115 virtual VideoSendStream* CreateVideoSendStream(
116 const VideoSendStream::Config& config, 116 VideoSendStream::Config config,
117 const VideoEncoderConfig& encoder_config) = 0; 117 VideoEncoderConfig encoder_config) = 0;
118 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; 118 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
119 119
120 virtual VideoReceiveStream* CreateVideoReceiveStream( 120 virtual VideoReceiveStream* CreateVideoReceiveStream(
121 VideoReceiveStream::Config configuration) = 0; 121 VideoReceiveStream::Config configuration) = 0;
122 virtual void DestroyVideoReceiveStream( 122 virtual void DestroyVideoReceiveStream(
123 VideoReceiveStream* receive_stream) = 0; 123 VideoReceiveStream* receive_stream) = 0;
124 124
125 // All received RTP and RTCP packets for the call should be inserted to this 125 // All received RTP and RTCP packets for the call should be inserted to this
126 // PacketReceiver. The PacketReceiver pointer is valid as long as the 126 // PacketReceiver. The PacketReceiver pointer is valid as long as the
127 // Call instance exists. 127 // Call instance exists.
(...skipping 26 matching lines...) Expand all
154 virtual bool StartEventLog(rtc::PlatformFile log_file, 154 virtual bool StartEventLog(rtc::PlatformFile log_file,
155 int64_t max_size_bytes) = 0; 155 int64_t max_size_bytes) = 0;
156 virtual void StopEventLog() = 0; 156 virtual void StopEventLog() = 0;
157 157
158 virtual ~Call() {} 158 virtual ~Call() {}
159 }; 159 };
160 160
161 } // namespace webrtc 161 } // namespace webrtc
162 162
163 #endif // WEBRTC_CALL_H_ 163 #endif // WEBRTC_CALL_H_
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