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Side by Side Diff: webrtc/modules/pacing/include/packet_router.h

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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_PACING_INCLUDE_PACKET_ROUTER_H_ 11 #ifndef WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
12 #define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_ 12 #define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
13 13
14 #include <list> 14 #include <map>
15 15
16 #include "webrtc/base/constructormagic.h" 16 #include "webrtc/base/constructormagic.h"
17 #include "webrtc/base/criticalsection.h"
17 #include "webrtc/base/scoped_ptr.h" 18 #include "webrtc/base/scoped_ptr.h"
18 #include "webrtc/base/thread_annotations.h" 19 #include "webrtc/base/thread_annotations.h"
19 #include "webrtc/common_types.h" 20 #include "webrtc/common_types.h"
21 #include "webrtc/modules/bitrate_controller/send_time_history.h"
stefan-webrtc 2015/07/22 10:49:00 Maybe this isn't the right place for the send_time
sprang_webrtc 2015/07/22 15:11:32 What would be a better place? modules/remote_bitra
stefan-webrtc 2015/07/27 12:13:49 I'm not sure... Maybe under pacing? We can keep it
sprang_webrtc 2015/07/28 13:29:27 I've move it around a few times now, trying to fit
20 #include "webrtc/modules/pacing/include/paced_sender.h" 22 #include "webrtc/modules/pacing/include/paced_sender.h"
23 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h"
24 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
21 25
22 namespace webrtc { 26 namespace webrtc {
23 27
24 class CriticalSectionWrapper;
25 class RTPFragmentationHeader;
26 class RtpRtcp; 28 class RtpRtcp;
27 struct RTPVideoHeader;
28 29
29 // PacketRouter routes outgoing data to the correct sending RTP module, based 30 // PacketRouter routes outgoing data to the correct sending RTP module, based
30 // on the simulcast layer in RTPVideoHeader. 31 // on the simulcast layer in RTPVideoHeader.
31 class PacketRouter : public PacedSender::Callback { 32 class PacketRouter : public PacedSender::Callback {
32 public: 33 public:
33 PacketRouter(); 34 PacketRouter();
34 virtual ~PacketRouter(); 35 virtual ~PacketRouter();
35 36
36 void AddRtpModule(RtpRtcp* rtp_module); 37 void AddRtpModule(RtpRtcp* rtp_module);
37 void RemoveRtpModule(RtpRtcp* rtp_module); 38 void RemoveRtpModule(RtpRtcp* rtp_module);
39 void OnSsrcChanged();
38 40
39 // Implements PacedSender::Callback. 41 // Implements PacedSender::Callback.
40 bool TimeToSendPacket(uint32_t ssrc, 42 bool TimeToSendPacket(uint32_t ssrc,
41 uint16_t sequence_number, 43 uint16_t sequence_number,
42 int64_t capture_timestamp, 44 int64_t capture_timestamp,
43 bool retransmission) override; 45 bool retransmission) override;
44 46
45 size_t TimeToSendPadding(size_t bytes) override; 47 size_t TimeToSendPadding(size_t bytes) override;
46 48
49 // Enable transport wide sequence number extension headers for outgoing
50 // RTP packets, specified by draft-holmer-rmcat-transport-wide-cc-extensions.
51 // This also enables send time history.
52 void EnableTransportWideFeedback();
53
54 void SetTransportWideSequenceNumber(uint16_t sequence_number);
55
56 // Populate PacketInfo.send_time_ms, by looking up the send time in the
57 // stored history index by sequence number. Returns the number of PacketInfo
58 // instances for which the lookup was successful.
59 size_t PopulateSendTimes(std::vector<PacketInfo>* packet_info);
60
61 uint16_t AllocateSequenceNumber();
62 void OnPacketSent(uint16_t sequence_number, int64_t send_time);
63
47 private: 64 private:
48 // TODO(holmer): When the new video API has launched, remove crit_ and 65 void UpdateModuleMap() EXCLUSIVE_LOCKS_REQUIRED(ssrc_lookup_lock_);
49 // assume rtp_modules_ will never change during a call. We should then also
50 // switch rtp_modules_ to a map from ssrc to rtp module.
51 rtc::scoped_ptr<CriticalSectionWrapper> crit_;
52 66
67 rtc::CriticalSection ssrc_lookup_lock_;
53 // Map from ssrc to sending rtp module. 68 // Map from ssrc to sending rtp module.
54 std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get()); 69 std::map<uint32_t, RtpRtcp*> rtp_modules_ GUARDED_BY(ssrc_lookup_lock_);
70 volatile int dirty_map_;
71
72 bool transport_wide_seq_enabled_;
73 volatile int transport_seq_;
74
75 rtc::CriticalSection history_lock_;
76 rtc::scoped_ptr<SendTimeHistory> send_time_history_ GUARDED_BY(history_lock_);
55 77
56 DISALLOW_COPY_AND_ASSIGN(PacketRouter); 78 DISALLOW_COPY_AND_ASSIGN(PacketRouter);
57 }; 79 };
58 } // namespace webrtc 80 } // namespace webrtc
59 #endif // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_ 81 #endif // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
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