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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Issue 2276833003: Change RtpSender::OnReceiveNACK name and signature (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: -one more <list> include Created 4 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
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
13 13
14 #include <list>
15 #include <set> 14 #include <set>
16 #include <utility> 15 #include <utility>
17 #include <vector> 16 #include <vector>
18 17
19 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/criticalsection.h"
20 #include "webrtc/base/gtest_prod_util.h" 19 #include "webrtc/base/gtest_prod_util.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
23 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h" 22 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
(...skipping 280 matching lines...) Expand 10 before | Expand all | Expand 10 after
305 uint32_t* nackRate) const override; 304 uint32_t* nackRate) const override;
306 305
307 // Good state of RTP receiver inform sender. 306 // Good state of RTP receiver inform sender.
308 int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) override; 307 int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) override;
309 308
310 void RegisterSendChannelRtpStatisticsCallback( 309 void RegisterSendChannelRtpStatisticsCallback(
311 StreamDataCountersCallback* callback) override; 310 StreamDataCountersCallback* callback) override;
312 StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback() 311 StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
313 const override; 312 const override;
314 313
315 void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers); 314 void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers);
316 void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks); 315 void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
317 316
318 void OnRequestSendReport(); 317 void OnRequestSendReport();
319 318
320 protected: 319 protected:
321 bool UpdateRTCPReceiveInformationTimers(); 320 bool UpdateRTCPReceiveInformationTimers();
322 321
323 RTPSender rtp_sender_; 322 RTPSender rtp_sender_;
324 323
325 RTCPSender rtcp_sender_; 324 RTCPSender rtcp_sender_;
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
360 PacketLossStats receive_loss_stats_; 359 PacketLossStats receive_loss_stats_;
361 360
362 // The processed RTT from RtcpRttStats. 361 // The processed RTT from RtcpRttStats.
363 rtc::CriticalSection critical_section_rtt_; 362 rtc::CriticalSection critical_section_rtt_;
364 int64_t rtt_ms_; 363 int64_t rtt_ms_;
365 }; 364 };
366 365
367 } // namespace webrtc 366 } // namespace webrtc
368 367
369 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 368 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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