Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(18)

Side by Side Diff: webrtc/video/payload_router.h

Issue 3002173002: Delete unneeded includes of atomic32.h. (Closed)
Patch Set: Created 3 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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
11 #ifndef WEBRTC_VIDEO_PAYLOAD_ROUTER_H_ 11 #ifndef WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
12 #define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_ 12 #define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
13 13
14 #include <vector> 14 #include <vector>
15 15
16 #include "webrtc/api/video_codecs/video_encoder.h" 16 #include "webrtc/api/video_codecs/video_encoder.h"
17 #include "webrtc/common_types.h" 17 #include "webrtc/common_types.h"
18 #include "webrtc/config.h" 18 #include "webrtc/config.h"
19 #include "webrtc/rtc_base/constructormagic.h" 19 #include "webrtc/rtc_base/constructormagic.h"
20 #include "webrtc/rtc_base/criticalsection.h" 20 #include "webrtc/rtc_base/criticalsection.h"
21 #include "webrtc/rtc_base/thread_annotations.h" 21 #include "webrtc/rtc_base/thread_annotations.h"
22 #include "webrtc/system_wrappers/include/atomic32.h"
23 22
24 namespace webrtc { 23 namespace webrtc {
25 24
26 class RTPFragmentationHeader; 25 class RTPFragmentationHeader;
27 class RtpRtcp; 26 class RtpRtcp;
28 struct RTPVideoHeader; 27 struct RTPVideoHeader;
29 28
30 // PayloadRouter routes outgoing data to the correct sending RTP module, based 29 // PayloadRouter routes outgoing data to the correct sending RTP module, based
31 // on the simulcast layer in RTPVideoHeader. 30 // on the simulcast layer in RTPVideoHeader.
32 class PayloadRouter : public EncodedImageCallback { 31 class PayloadRouter : public EncodedImageCallback {
(...skipping 26 matching lines...) Expand all
59 // Rtp modules are assumed to be sorted in simulcast index order. Not owned. 58 // Rtp modules are assumed to be sorted in simulcast index order. Not owned.
60 const std::vector<RtpRtcp*> rtp_modules_; 59 const std::vector<RtpRtcp*> rtp_modules_;
61 const int payload_type_; 60 const int payload_type_;
62 61
63 RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter); 62 RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
64 }; 63 };
65 64
66 } // namespace webrtc 65 } // namespace webrtc
67 66
68 #endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_ 67 #endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698