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Side by Side Diff: webrtc/modules/pacing/packet_router.cc

Issue 1628683002: Use separate rtp module lists for send and receive in PacketRouter. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 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 #include "webrtc/modules/pacing/packet_router.h" 11 #include "webrtc/modules/pacing/packet_router.h"
12 12
13 #include "webrtc/base/atomicops.h" 13 #include "webrtc/base/atomicops.h"
14 #include "webrtc/base/checks.h" 14 #include "webrtc/base/checks.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
18 18
19 namespace webrtc { 19 namespace webrtc {
20 20
21 namespace {
22 void AddModule(RtpRtcp* rtp_module, std::list<RtpRtcp*>* rtp_modules) {
23 RTC_DCHECK(std::find(rtp_modules->begin(), rtp_modules->end(), rtp_module) ==
24 rtp_modules->end());
25 rtp_modules->push_back(rtp_module);
26 }
27
28 bool RemoveModule(RtpRtcp* rtp_module, std::list<RtpRtcp*>* rtp_modules) {
29 auto it = std::find(rtp_modules->begin(), rtp_modules->end(), rtp_module);
30 if (it == rtp_modules->end())
31 return false;
32 rtp_modules->erase(it);
tommi 2016/01/23 17:23:26 if we didn't have to search the receiver list when
stefan-webrtc 2016/01/23 17:46:03 Right, that would be an option, and I can definite
tommi 2016/01/24 10:30:08 We can use std::find() inside an RTC_DCHECK to che
stefan-webrtc 2016/01/25 11:54:46 I think I'll just add a bool sender to RemoveRtpMo
33 return true;
34 }
35
36 bool SendFeedback(rtcp::TransportFeedback* packet,
37 std::list<RtpRtcp*>* rtp_modules) {
38 for (auto* rtp_module : *rtp_modules) {
39 packet->WithPacketSenderSsrc(rtp_module->SSRC());
40 if (rtp_module->SendFeedbackPacket(*packet))
41 return true;
42 }
43 return false;
44 }
45 }
46
21 PacketRouter::PacketRouter() : transport_seq_(0) { 47 PacketRouter::PacketRouter() : transport_seq_(0) {
22 } 48 }
23 49
24 PacketRouter::~PacketRouter() { 50 PacketRouter::~PacketRouter() {
25 RTC_DCHECK(rtp_modules_.empty()); 51 RTC_DCHECK(send_rtp_modules_.empty());
52 RTC_DCHECK(recv_rtp_modules_.empty());
26 } 53 }
27 54
28 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) { 55 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module, bool sender) {
29 rtc::CritScope cs(&modules_lock_); 56 rtc::CritScope cs(&modules_lock_);
30 RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) == 57
31 rtp_modules_.end()); 58 if (sender) {
tommi 2016/01/23 17:23:26 is it good to lock both sender and receiver maps w
stefan-webrtc 2016/01/23 17:46:03 We could have different locks, but I don't think i
tommi 2016/01/24 10:30:08 I think that this particular class is very light b
stefan-webrtc 2016/01/25 11:54:46 Agree it's not very clear how expensive it is to s
32 rtp_modules_.push_back(rtp_module); 59 AddModule(rtp_module, &send_rtp_modules_);
60 } else {
61 AddModule(rtp_module, &recv_rtp_modules_);
62 }
33 } 63 }
34 64
35 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) { 65 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
36 rtc::CritScope cs(&modules_lock_); 66 rtc::CritScope cs(&modules_lock_);
37 auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module); 67 bool removed = RemoveModule(rtp_module, &send_rtp_modules_);
38 RTC_DCHECK(it != rtp_modules_.end()); 68 if (!removed)
39 rtp_modules_.erase(it); 69 removed = RemoveModule(rtp_module, &recv_rtp_modules_);
70 RTC_DCHECK(removed);
40 } 71 }
41 72
42 bool PacketRouter::TimeToSendPacket(uint32_t ssrc, 73 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
43 uint16_t sequence_number, 74 uint16_t sequence_number,
44 int64_t capture_timestamp, 75 int64_t capture_timestamp,
45 bool retransmission) { 76 bool retransmission) {
46 rtc::CritScope cs(&modules_lock_); 77 rtc::CritScope cs(&modules_lock_);
47 for (auto* rtp_module : rtp_modules_) { 78 for (auto* rtp_module : send_rtp_modules_) {
48 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) { 79 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
49 return rtp_module->TimeToSendPacket(ssrc, sequence_number, 80 return rtp_module->TimeToSendPacket(ssrc, sequence_number,
50 capture_timestamp, retransmission); 81 capture_timestamp, retransmission);
51 } 82 }
52 } 83 }
53 return true; 84 return true;
54 } 85 }
55 86
56 size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) { 87 size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
57 size_t total_bytes_sent = 0; 88 size_t total_bytes_sent = 0;
58 rtc::CritScope cs(&modules_lock_); 89 rtc::CritScope cs(&modules_lock_);
59 for (RtpRtcp* module : rtp_modules_) { 90 for (RtpRtcp* module : send_rtp_modules_) {
60 if (module->SendingMedia()) { 91 if (module->SendingMedia()) {
61 size_t bytes_sent = 92 size_t bytes_sent =
62 module->TimeToSendPadding(bytes_to_send - total_bytes_sent); 93 module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
63 total_bytes_sent += bytes_sent; 94 total_bytes_sent += bytes_sent;
64 if (total_bytes_sent >= bytes_to_send) 95 if (total_bytes_sent >= bytes_to_send)
65 break; 96 break;
66 } 97 }
67 } 98 }
68 return total_bytes_sent; 99 return total_bytes_sent;
69 } 100 }
(...skipping 15 matching lines...) Expand all
85 // return value saves us a load on retry. 116 // return value saves us a load on retry.
86 prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq, 117 prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
87 new_seq); 118 new_seq);
88 } while (prev_seq != desired_prev_seq); 119 } while (prev_seq != desired_prev_seq);
89 120
90 return new_seq; 121 return new_seq;
91 } 122 }
92 123
93 bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) { 124 bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
94 rtc::CritScope cs(&modules_lock_); 125 rtc::CritScope cs(&modules_lock_);
95 for (auto* rtp_module : rtp_modules_) { 126 if (!::webrtc::SendFeedback(packet, &recv_rtp_modules_))
96 packet->WithPacketSenderSsrc(rtp_module->SSRC()); 127 return ::webrtc::SendFeedback(packet, &send_rtp_modules_);
97 if (rtp_module->SendFeedbackPacket(*packet)) 128 return true;
tommi 2016/01/23 17:23:25 nit: return webrtc::SendFeedback(packet, &recv_rt
stefan-webrtc 2016/01/23 17:46:04 Agree, I think it would be better to find the modu
tommi 2016/01/24 10:30:08 No chance that it could be removed while the call
stefan-webrtc 2016/01/25 11:54:46 I thought I could guarantee that, but unfortunatel
98 return true;
99 }
100 return false;
101 } 129 }
102 130
103 } // namespace webrtc 131 } // namespace webrtc
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