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

Issue 1362923002: Revert of Wire up send-side bandwidth estimation. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Manual merge of revert Created 5 years, 3 months ago
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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_receiver_help.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
12 12
13 #include <assert.h> // assert 13 #include <assert.h> // assert
14 #include <string.h> // memset 14 #include <string.h> // memset
15 15
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
18 17
19 namespace webrtc { 18 namespace webrtc {
20 using namespace RTCPHelp; 19 using namespace RTCPHelp;
21 20
22 RTCPPacketInformation::RTCPPacketInformation() 21 RTCPPacketInformation::RTCPPacketInformation()
23 : rtcpPacketTypeFlags(0), 22 : rtcpPacketTypeFlags(0),
24 remoteSSRC(0), 23 remoteSSRC(0),
25 nackSequenceNumbers(), 24 nackSequenceNumbers(),
26 applicationSubType(0), 25 applicationSubType(0),
27 applicationName(0), 26 applicationName(0),
28 applicationData(), 27 applicationData(),
29 applicationLength(0), 28 applicationLength(0),
30 rtt(0), 29 rtt(0),
31 interArrivalJitter(0), 30 interArrivalJitter(0),
32 sliPictureId(0), 31 sliPictureId(0),
33 rpsiPictureId(0), 32 rpsiPictureId(0),
34 receiverEstimatedMaxBitrate(0), 33 receiverEstimatedMaxBitrate(0),
35 ntp_secs(0), 34 ntp_secs(0),
36 ntp_frac(0), 35 ntp_frac(0),
37 rtp_timestamp(0), 36 rtp_timestamp(0),
38 xr_originator_ssrc(0), 37 xr_originator_ssrc(0),
39 xr_dlrr_item(false), 38 xr_dlrr_item(false),
40 VoIPMetric(nullptr) {} 39 VoIPMetric(NULL) {
40 }
41 41
42 RTCPPacketInformation::~RTCPPacketInformation() 42 RTCPPacketInformation::~RTCPPacketInformation()
43 { 43 {
44 delete [] applicationData; 44 delete [] applicationData;
45 delete VoIPMetric; 45 delete VoIPMetric;
46 } 46 }
47 47
48 void 48 void
49 RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric) 49 RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
50 { 50 {
(...skipping 133 matching lines...) Expand 10 before | Expand all | Expand 10 after
184 TmmbrSet.PacketOH(sourceIdx), 184 TmmbrSet.PacketOH(sourceIdx),
185 TmmbrSet.Ssrc(sourceIdx)); 185 TmmbrSet.Ssrc(sourceIdx));
186 return 0; 186 return 0;
187 } 187 }
188 188
189 void RTCPReceiveInformation::VerifyAndAllocateBoundingSet( 189 void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
190 const uint32_t minimumSize) { 190 const uint32_t minimumSize) {
191 TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize); 191 TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
192 } 192 }
193 } // namespace webrtc 193 } // namespace webrtc
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