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Side by Side Diff: webrtc/modules/pacing/packet_router.cc

Issue 1290813008: Add RemoteEstimatorProxy for capturing receive times (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 4 months ago
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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 #include "webrtc/modules/pacing/include/packet_router.h" 11 #include "webrtc/modules/pacing/include/packet_router.h"
12 12
13 #include "webrtc/base/atomicops.h" 13 #include "webrtc/base/atomicops.h"
14 #include "webrtc/base/checks.h" 14 #include "webrtc/base/checks.h"
15 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 15 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
16 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 16 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
17 18
18 namespace webrtc { 19 namespace webrtc {
19 20
20 PacketRouter::PacketRouter() : transport_seq_(0) { 21 PacketRouter::PacketRouter() : transport_seq_(0) {
21 } 22 }
22 23
23 PacketRouter::~PacketRouter() { 24 PacketRouter::~PacketRouter() {
24 DCHECK(rtp_modules_.empty()); 25 DCHECK(rtp_modules_.empty());
25 } 26 }
26 27
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82 // time the CAS operation was executed. Thus, if prev_seq is returned, the 83 // time the CAS operation was executed. Thus, if prev_seq is returned, the
83 // operation was successful - otherwise we need to retry. Saving the 84 // operation was successful - otherwise we need to retry. Saving the
84 // return value saves us a load on retry. 85 // return value saves us a load on retry.
85 prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq, 86 prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
86 new_seq); 87 new_seq);
87 } while (prev_seq != desired_prev_seq); 88 } while (prev_seq != desired_prev_seq);
88 89
89 return new_seq; 90 return new_seq;
90 } 91 }
91 92
93 bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
94 rtc::CritScope cs(&modules_lock_);
95 for (auto* rtp_module : rtp_modules_) {
96 packet->WithPacketSenderSsrc(rtp_module->SSRC());
stefan-webrtc 2015/09/04 07:31:36 Does this replace the ssrc in the feedback packet?
sprang_webrtc 2015/09/04 13:29:44 The packet sender ssrc is changed, yes. There is a
97 if (rtp_module->SendFeedbackPacket(*packet))
98 return true;
99 }
100 return false;
101 }
102
92 } // namespace webrtc 103 } // namespace webrtc
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