Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(5)

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

Issue 2528993002: Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (Closed)
Patch Set: Created 4 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 253 matching lines...) Expand 10 before | Expand all | Expand 10 after
264 } 264 }
265 } 265 }
266 266
267 private: 267 private:
268 class Handler : public RtpStreamInterface { 268 class Handler : public RtpStreamInterface {
269 public: 269 public:
270 Handler(uint32_t ssrc, 270 Handler(uint32_t ssrc,
271 const PayloadTypes& payload_types, 271 const PayloadTypes& payload_types,
272 LostPackets* lost_packets) 272 LostPackets* lost_packets)
273 : rtp_header_parser_(RtpHeaderParser::Create()), 273 : rtp_header_parser_(RtpHeaderParser::Create()),
274 rtp_payload_registry_(new RTPPayloadRegistry()), 274 rtp_payload_registry_(new RTPPayloadRegistry(
275 RTPPayloadStrategy::CreateStrategy(false))),
275 rtp_module_(), 276 rtp_module_(),
276 payload_sink_(), 277 payload_sink_(),
277 ssrc_(ssrc), 278 ssrc_(ssrc),
278 payload_types_(payload_types), 279 payload_types_(payload_types),
279 lost_packets_(lost_packets) { 280 lost_packets_(lost_packets) {
280 assert(lost_packets); 281 assert(lost_packets);
281 } 282 }
282 virtual ~Handler() {} 283 virtual ~Handler() {}
283 284
284 virtual void ResendPackets(const uint16_t* sequence_numbers, 285 virtual void ResendPackets(const uint16_t* sequence_numbers,
(...skipping 199 matching lines...) Expand 10 before | Expand all | Expand 10 after
484 } 485 }
485 } 486 }
486 487
487 std::unique_ptr<RtpPlayerImpl> impl( 488 std::unique_ptr<RtpPlayerImpl> impl(
488 new RtpPlayerImpl(payload_sink_factory, payload_types, clock, 489 new RtpPlayerImpl(payload_sink_factory, payload_types, clock,
489 &packet_source, loss_rate, rtt_ms, reordering)); 490 &packet_source, loss_rate, rtt_ms, reordering));
490 return impl.release(); 491 return impl.release();
491 } 492 }
492 } // namespace rtpplayer 493 } // namespace rtpplayer
493 } // namespace webrtc 494 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc ('k') | webrtc/video/rtp_stream_receiver.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698