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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/remote_bitrate_estimator/remote_estimator_proxy.h" | 11 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h" |
12 | 12 |
13 #include <limits> | 13 #include <limits> |
14 #include <algorithm> | |
14 | 15 |
15 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
16 #include "webrtc/base/logging.h" | 17 #include "webrtc/base/logging.h" |
17 #include "webrtc/system_wrappers/include/clock.h" | 18 #include "webrtc/system_wrappers/include/clock.h" |
19 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | |
18 #include "webrtc/modules/pacing/packet_router.h" | 20 #include "webrtc/modules/pacing/packet_router.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
21 | 23 |
22 namespace webrtc { | 24 namespace webrtc { |
23 | 25 |
24 // TODO(sprang): Tune these! | 26 // TODO(sprang): Tune these! |
25 const int RemoteEstimatorProxy::kDefaultProcessIntervalMs = 50; | 27 const int RemoteEstimatorProxy::kMinPacketCountRecievedBitrate = 50; |
26 const int RemoteEstimatorProxy::kBackWindowMs = 500; | 28 const int RemoteEstimatorProxy::kBackWindowMs = 500; |
29 const int RemoteEstimatorProxy::kMinSendIntervalMs = 50; | |
30 const int RemoteEstimatorProxy::kMaxSendIntervalMs = 250; | |
27 | 31 |
28 // The maximum allowed value for a timestamp in milliseconds. This is lower | 32 // The maximum allowed value for a timestamp in milliseconds. This is lower |
29 // than the numerical limit since we often convert to microseconds. | 33 // than the numerical limit since we often convert to microseconds. |
30 static constexpr int64_t kMaxTimeMs = | 34 static constexpr int64_t kMaxTimeMs = |
31 std::numeric_limits<int64_t>::max() / 1000; | 35 std::numeric_limits<int64_t>::max() / 1000; |
32 | 36 |
33 RemoteEstimatorProxy::RemoteEstimatorProxy(Clock* clock, | 37 RemoteEstimatorProxy::RemoteEstimatorProxy( |
34 PacketRouter* packet_router) | 38 Clock* clock, |
39 PacketRouter* packet_router, | |
40 BitrateController* bitrate_controller) | |
35 : clock_(clock), | 41 : clock_(clock), |
36 packet_router_(packet_router), | 42 packet_router_(packet_router), |
37 last_process_time_ms_(-1), | 43 last_process_time_ms_(-1), |
38 media_ssrc_(0), | 44 media_ssrc_(0), |
39 feedback_sequence_(0), | 45 feedback_sequence_(0), |
40 window_start_seq_(-1) {} | 46 window_start_seq_(-1), |
47 bitrate_controller_(bitrate_controller) {} | |
41 | 48 |
42 RemoteEstimatorProxy::~RemoteEstimatorProxy() {} | 49 RemoteEstimatorProxy::~RemoteEstimatorProxy() {} |
43 | 50 |
44 void RemoteEstimatorProxy::IncomingPacketFeedbackVector( | 51 void RemoteEstimatorProxy::IncomingPacketFeedbackVector( |
45 const std::vector<PacketInfo>& packet_feedback_vector) { | 52 const std::vector<PacketInfo>& packet_feedback_vector) { |
46 rtc::CritScope cs(&lock_); | 53 rtc::CritScope cs(&lock_); |
47 for (PacketInfo info : packet_feedback_vector) | 54 for (PacketInfo info : packet_feedback_vector) |
48 OnPacketArrival(info.sequence_number, info.arrival_time_ms); | 55 OnPacketArrival(info.sequence_number, info.arrival_time_ms); |
49 } | 56 } |
50 | 57 |
51 void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms, | 58 void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms, |
52 size_t payload_size, | 59 size_t payload_size, |
53 const RTPHeader& header) { | 60 const RTPHeader& header) { |
54 if (!header.extension.hasTransportSequenceNumber) { | 61 if (!header.extension.hasTransportSequenceNumber) { |
55 LOG(LS_WARNING) << "RemoteEstimatorProxy: Incoming packet " | 62 LOG(LS_WARNING) << "RemoteEstimatorProxy: Incoming packet " |
56 "is missing the transport sequence number extension!"; | 63 "is missing the transport sequence number extension!"; |
57 return; | 64 return; |
58 } | 65 } |
59 rtc::CritScope cs(&lock_); | 66 rtc::CritScope cs(&lock_); |
60 media_ssrc_ = header.ssrc; | 67 media_ssrc_ = header.ssrc; |
61 | 68 |
62 OnPacketArrival(header.extension.transportSequenceNumber, arrival_time_ms); | 69 OnPacketArrival(header.extension.transportSequenceNumber, arrival_time_ms); |
63 } | 70 } |
64 | 71 |
65 bool RemoteEstimatorProxy::LatestEstimate(std::vector<unsigned int>* ssrcs, | 72 bool RemoteEstimatorProxy::LatestEstimate(std::vector<unsigned int>* ssrcs, |
66 unsigned int* bitrate_bps) const { | 73 unsigned int* bitrate_bps) const { |
67 return false; | 74 return false; |
68 } | 75 } |
69 | 76 |
70 int64_t RemoteEstimatorProxy::TimeUntilNextProcess() { | 77 int64_t RemoteEstimatorProxy::TimeUntilNextProcess() { |
minyue-webrtc
2016/10/26 10:11:35
general question: should we update send_interval_m
michaelt
2016/11/03 10:20:20
Since send_interval_ms is just dependent on the ac
minyue-webrtc
2016/11/03 10:39:38
That reads much nicer to me.
| |
71 int64_t now = clock_->TimeInMilliseconds(); | 78 int64_t now = clock_->TimeInMilliseconds(); |
72 int64_t time_until_next = 0; | 79 int64_t time_until_next = 0; |
73 if (last_process_time_ms_ != -1 && | 80 if (last_process_time_ms_ != -1) { |
74 now - last_process_time_ms_ < kDefaultProcessIntervalMs) { | 81 uint32_t bitrate = 0; |
75 time_until_next = (last_process_time_ms_ + kDefaultProcessIntervalMs - now); | 82 bitrate_controller_->AvailableBandwidth(&bitrate); |
83 // Ipv4(20B) + UDP(8B) + SRTP(4B) + AverageTwccReport(20B) | |
84 constexpr int kTwccReportSize = 20 + 8 + 4 + 20; | |
85 int send_interval_ms = | |
86 kTwccReportSize * 8.0 * 1000.0 / (0.05 * bitrate) + 0.5; | |
minyue-webrtc
2016/10/26 10:11:35
Would you add a comment, something like:
"Let TWC
michaelt
2016/11/03 10:20:20
Done.
| |
87 send_interval_ms = std::min(std::max(send_interval_ms, kMinSendIntervalMs), | |
88 kMaxSendIntervalMs); | |
89 if (now - last_process_time_ms_ < send_interval_ms) | |
90 time_until_next = (last_process_time_ms_ + send_interval_ms - now); | |
76 } | 91 } |
77 return time_until_next; | 92 return time_until_next; |
78 } | 93 } |
79 | 94 |
80 void RemoteEstimatorProxy::Process() { | 95 void RemoteEstimatorProxy::Process() { |
81 last_process_time_ms_ = clock_->TimeInMilliseconds(); | 96 last_process_time_ms_ = clock_->TimeInMilliseconds(); |
82 | 97 |
83 bool more_to_build = true; | 98 bool more_to_build = true; |
84 while (more_to_build) { | 99 while (more_to_build) { |
85 rtcp::TransportFeedback feedback_packet; | 100 rtcp::TransportFeedback feedback_packet; |
(...skipping 88 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
174 // Note: Don't erase items from packet_arrival_times_ after sending, in case | 189 // Note: Don't erase items from packet_arrival_times_ after sending, in case |
175 // they need to be re-sent after a reordering. Removal will be handled | 190 // they need to be re-sent after a reordering. Removal will be handled |
176 // by OnPacketArrival once packets are too old. | 191 // by OnPacketArrival once packets are too old. |
177 window_start_seq_ = it->first + 1; | 192 window_start_seq_ = it->first + 1; |
178 } | 193 } |
179 | 194 |
180 return true; | 195 return true; |
181 } | 196 } |
182 | 197 |
183 } // namespace webrtc | 198 } // namespace webrtc |
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