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Issue 1779773002: Make rtc::TimestampWrapAroundHandler handle backwards wrapping (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added unit test for "no negative start" Created 4 years, 9 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 #include <stdio.h> 10 #include <stdio.h>
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154 const PacketOptions& options) override { 154 const PacketOptions& options) override {
155 RtpUtility::RtpHeaderParser parser(packet, length); 155 RtpUtility::RtpHeaderParser parser(packet, length);
156 RTPHeader header; 156 RTPHeader header;
157 parser.Parse(&header); 157 parser.Parse(&header);
158 158
159 int64_t current_time = 159 int64_t current_time =
160 Clock::GetRealTimeClock()->CurrentNtpInMilliseconds(); 160 Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
161 bool result = transport_->SendRtp(packet, length, options); 161 bool result = transport_->SendRtp(packet, length, options);
162 { 162 {
163 rtc::CritScope lock(&crit_); 163 rtc::CritScope lock(&crit_);
164 int64_t timestamp = wrap_handler_.Unwrap(header.timestamp);
165 164
166 if (rtp_timestamp_delta_ == 0) { 165 if (rtp_timestamp_delta_ == 0) {
167 rtp_timestamp_delta_ = timestamp - first_send_frame_.timestamp(); 166 rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
168 first_send_frame_.Reset(); 167 first_send_frame_.Reset();
169 } 168 }
170 timestamp -= rtp_timestamp_delta_; 169 int64_t timestamp =
170 wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
171 send_times_[timestamp] = current_time; 171 send_times_[timestamp] = current_time;
172 if (!transport_->DiscardedLastPacket() && 172 if (!transport_->DiscardedLastPacket() &&
173 header.ssrc == ssrc_to_analyze_) { 173 header.ssrc == ssrc_to_analyze_) {
174 encoded_frame_sizes_[timestamp] += 174 encoded_frame_sizes_[timestamp] +=
175 length - (header.headerLength + header.paddingLength); 175 length - (header.headerLength + header.paddingLength);
176 } 176 }
177 } 177 }
178 return result; 178 return result;
179 } 179 }
180 180
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1095 video_send_stream_->Stop(); 1095 video_send_stream_->Stop();
1096 receive_stream->Stop(); 1096 receive_stream->Stop();
1097 1097
1098 call->DestroyVideoReceiveStream(receive_stream); 1098 call->DestroyVideoReceiveStream(receive_stream);
1099 call->DestroyVideoSendStream(video_send_stream_); 1099 call->DestroyVideoSendStream(video_send_stream_);
1100 1100
1101 transport.StopSending(); 1101 transport.StopSending();
1102 } 1102 }
1103 1103
1104 } // namespace webrtc 1104 } // namespace webrtc
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