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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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