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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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