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Side by Side Diff: webrtc/video/rtp_stream_receiver.cc

Issue 2064523002: GN: Add video_engine_tests (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: audio_receive_stream_unittest compile Created 4 years, 6 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
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219 int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const { 219 int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const {
220 return rtp_receiver_->CSRCs(csrcs); 220 return rtp_receiver_->CSRCs(csrcs);
221 } 221 }
222 222
223 RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const { 223 RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const {
224 return rtp_receiver_.get(); 224 return rtp_receiver_.get();
225 } 225 }
226 226
227 int32_t RtpStreamReceiver::OnReceivedPayloadData( 227 int32_t RtpStreamReceiver::OnReceivedPayloadData(
228 const uint8_t* payload_data, 228 const uint8_t* payload_data,
229 const size_t payload_size, 229 size_t payload_size,
230 const WebRtcRTPHeader* rtp_header) { 230 const WebRtcRTPHeader* rtp_header) {
231 RTC_DCHECK(video_receiver_); 231 RTC_DCHECK(video_receiver_);
232 WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; 232 WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
233 rtp_header_with_ntp.ntp_time_ms = 233 rtp_header_with_ntp.ntp_time_ms =
234 ntp_estimator_.Estimate(rtp_header->header.timestamp); 234 ntp_estimator_.Estimate(rtp_header->header.timestamp);
235 if (video_receiver_->IncomingPacket(payload_data, payload_size, 235 if (video_receiver_->IncomingPacket(payload_data, payload_size,
236 rtp_header_with_ntp) != 0) { 236 rtp_header_with_ntp) != 0) {
237 // Check this... 237 // Check this...
238 return -1; 238 return -1;
239 } 239 }
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536 const std::string& extension, int id) { 536 const std::string& extension, int id) {
537 // One-byte-extension local identifiers are in the range 1-14 inclusive. 537 // One-byte-extension local identifiers are in the range 1-14 inclusive.
538 RTC_DCHECK_GE(id, 1); 538 RTC_DCHECK_GE(id, 1);
539 RTC_DCHECK_LE(id, 14); 539 RTC_DCHECK_LE(id, 14);
540 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); 540 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
541 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( 541 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
542 StringToRtpExtensionType(extension), id)); 542 StringToRtpExtensionType(extension), id));
543 } 543 }
544 544
545 } // namespace webrtc 545 } // namespace webrtc
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