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Side by Side Diff: webrtc/modules/video_coding/main/test/rtp_player.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase 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
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210 210
211 rtc::scoped_ptr<Handler> handler( 211 rtc::scoped_ptr<Handler> handler(
212 new Handler(ssrc, payload_types_, lost_packets)); 212 new Handler(ssrc, payload_types_, lost_packets));
213 handler->payload_sink_.reset(payload_sink_factory_->Create(handler.get())); 213 handler->payload_sink_.reset(payload_sink_factory_->Create(handler.get()));
214 if (handler->payload_sink_.get() == NULL) { 214 if (handler->payload_sink_.get() == NULL) {
215 return -1; 215 return -1;
216 } 216 }
217 217
218 RtpRtcp::Configuration configuration; 218 RtpRtcp::Configuration configuration;
219 configuration.clock = clock; 219 configuration.clock = clock;
220 configuration.id = 1;
221 configuration.audio = false; 220 configuration.audio = false;
222 handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver( 221 handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver(
223 configuration.id, configuration.clock, handler->payload_sink_.get(), 222 configuration.clock, handler->payload_sink_.get(), NULL,
224 NULL, handler->rtp_payload_registry_.get())); 223 handler->rtp_payload_registry_.get()));
225 if (handler->rtp_module_.get() == NULL) { 224 if (handler->rtp_module_.get() == NULL) {
226 return -1; 225 return -1;
227 } 226 }
228 227
229 handler->rtp_module_->SetNACKStatus(kNackOff); 228 handler->rtp_module_->SetNACKStatus(kNackOff);
230 handler->rtp_header_parser_->RegisterRtpHeaderExtension( 229 handler->rtp_header_parser_->RegisterRtpHeaderExtension(
231 kRtpExtensionTransmissionTimeOffset, 230 kRtpExtensionTransmissionTimeOffset,
232 kDefaultTransmissionTimeOffsetExtensionId); 231 kDefaultTransmissionTimeOffsetExtensionId);
233 232
234 for (PayloadTypesIterator it = payload_types_.begin(); 233 for (PayloadTypesIterator it = payload_types_.begin();
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485 } 484 }
486 } 485 }
487 486
488 rtc::scoped_ptr<RtpPlayerImpl> impl( 487 rtc::scoped_ptr<RtpPlayerImpl> impl(
489 new RtpPlayerImpl(payload_sink_factory, payload_types, clock, 488 new RtpPlayerImpl(payload_sink_factory, payload_types, clock,
490 &packet_source, loss_rate, rtt_ms, reordering)); 489 &packet_source, loss_rate, rtt_ms, reordering));
491 return impl.release(); 490 return impl.release();
492 } 491 }
493 } // namespace rtpplayer 492 } // namespace rtpplayer
494 } // namespace webrtc 493 } // namespace webrtc
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