Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1426)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

Issue 1430013003: rtcp::Bye packet moved to own file and got a Parse function (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month 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 | « webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc ('k') | no next file » | 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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
12 12
13 #include <assert.h> // assert 13 #include <assert.h> // assert
14 #include <stdlib.h> // rand 14 #include <stdlib.h> // rand
15 #include <string.h> // memcpy 15 #include <string.h> // memcpy
16 16
17 #include <algorithm> // min 17 #include <algorithm> // min
18 #include <limits> // max 18 #include <limits> // max
19 19
20 #include "webrtc/base/checks.h" 20 #include "webrtc/base/checks.h"
21 #include "webrtc/base/logging.h" 21 #include "webrtc/base/logging.h"
22 #include "webrtc/base/trace_event.h" 22 #include "webrtc/base/trace_event.h"
23 #include "webrtc/common_types.h" 23 #include "webrtc/common_types.h"
24 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 24 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" 27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
27 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" 28 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 29 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
29 30
30 namespace webrtc { 31 namespace webrtc {
31 32
32 using RTCPUtility::RTCPCnameInformation; 33 using RTCPUtility::RTCPCnameInformation;
33 34
34 NACKStringBuilder::NACKStringBuilder() 35 NACKStringBuilder::NACKStringBuilder()
35 : stream_(""), count_(0), prevNack_(0), consecutive_(false) { 36 : stream_(""), count_(0), prevNack_(0), consecutive_(false) {
(...skipping 1184 matching lines...) Expand 10 before | Expand all | Expand 10 after
1220 Transport* const transport_; 1221 Transport* const transport_;
1221 bool send_failure_; 1222 bool send_failure_;
1222 } sender(transport_); 1223 } sender(transport_);
1223 1224
1224 uint8_t buffer[IP_PACKET_SIZE]; 1225 uint8_t buffer[IP_PACKET_SIZE];
1225 return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) && 1226 return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
1226 !sender.send_failure_; 1227 !sender.send_failure_;
1227 } 1228 }
1228 1229
1229 } // namespace webrtc 1230 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698