1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
|
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include <algorithm>
#include <cstdint>
#include <cstdio>
#include <memory>
#include <string>
#include <utility>
#include "absl/strings/match.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/units/time_delta.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds(2);
constexpr TimeDelta kSendTimeGroupLength = TimeDelta::Millis(5);
// This ssrc is used to fulfill the current API but will be removed
// after the API has been changed.
constexpr uint32_t kFixedSsrc = 0;
} // namespace
constexpr char BweSeparateAudioPacketsSettings::kKey[];
BweSeparateAudioPacketsSettings::BweSeparateAudioPacketsSettings(
const FieldTrialsView* key_value_config) {
Parser()->Parse(
key_value_config->Lookup(BweSeparateAudioPacketsSettings::kKey));
}
std::unique_ptr<StructParametersParser>
BweSeparateAudioPacketsSettings::Parser() {
return StructParametersParser::Create( //
"enabled", &enabled, //
"packet_threshold", &packet_threshold, //
"time_threshold", &time_threshold);
}
DelayBasedBwe::Result::Result()
: updated(false),
probe(false),
target_bitrate(DataRate::Zero()),
recovered_from_overuse(false) {}
DelayBasedBwe::DelayBasedBwe(const FieldTrialsView* key_value_config,
RtcEventLog* event_log,
NetworkStatePredictor* network_state_predictor)
: event_log_(event_log),
key_value_config_(key_value_config),
separate_audio_(key_value_config),
audio_packets_since_last_video_(0),
last_video_packet_recv_time_(Timestamp::MinusInfinity()),
network_state_predictor_(network_state_predictor),
video_delay_detector_(
new TrendlineEstimator(key_value_config_, network_state_predictor_)),
audio_delay_detector_(
new TrendlineEstimator(key_value_config_, network_state_predictor_)),
active_delay_detector_(video_delay_detector_.get()),
last_seen_packet_(Timestamp::MinusInfinity()),
uma_recorded_(false),
rate_control_(key_value_config, /*send_side=*/true),
prev_bitrate_(DataRate::Zero()),
prev_state_(BandwidthUsage::kBwNormal) {
RTC_LOG(LS_INFO)
<< "Initialized DelayBasedBwe with separate audio overuse detection"
<< separate_audio_.Parser()->Encode();
}
DelayBasedBwe::~DelayBasedBwe() {}
DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
const TransportPacketsFeedback& msg,
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
absl::optional<NetworkStateEstimate> network_estimate,
bool in_alr) {
RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
auto packet_feedback_vector = msg.SortedByReceiveTime();
// TODO(holmer): An empty feedback vector here likely means that
// all acks were too late and that the send time history had
// timed out. We should reduce the rate when this occurs.
if (packet_feedback_vector.empty()) {
RTC_LOG(LS_WARNING) << "Very late feedback received.";
return DelayBasedBwe::Result();
}
if (!uma_recorded_) {
RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
BweNames::kSendSideTransportSeqNum,
BweNames::kBweNamesMax);
uma_recorded_ = true;
}
bool delayed_feedback = true;
bool recovered_from_overuse = false;
BandwidthUsage prev_detector_state = active_delay_detector_->State();
for (const auto& packet_feedback : packet_feedback_vector) {
delayed_feedback = false;
IncomingPacketFeedback(packet_feedback, msg.feedback_time);
if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
active_delay_detector_->State() == BandwidthUsage::kBwNormal) {
recovered_from_overuse = true;
}
prev_detector_state = active_delay_detector_->State();
}
if (delayed_feedback) {
// TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
// against building very large network queues.
return Result();
}
rate_control_.SetInApplicationLimitedRegion(in_alr);
rate_control_.SetNetworkStateEstimate(network_estimate);
return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
std::move(network_estimate),
recovered_from_overuse, in_alr, msg.feedback_time);
}
void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
Timestamp at_time) {
// Reset if the stream has timed out.
if (last_seen_packet_.IsInfinite() ||
at_time - last_seen_packet_ > kStreamTimeOut) {
video_inter_arrival_delta_ =
std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
audio_inter_arrival_delta_ =
std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
video_delay_detector_.reset(
new TrendlineEstimator(key_value_config_, network_state_predictor_));
audio_delay_detector_.reset(
new TrendlineEstimator(key_value_config_, network_state_predictor_));
active_delay_detector_ = video_delay_detector_.get();
}
last_seen_packet_ = at_time;
// As an alternative to ignoring small packets, we can separate audio and
// video packets for overuse detection.
DelayIncreaseDetectorInterface* delay_detector_for_packet =
video_delay_detector_.get();
if (separate_audio_.enabled) {
if (packet_feedback.sent_packet.audio) {
delay_detector_for_packet = audio_delay_detector_.get();
audio_packets_since_last_video_++;
if (audio_packets_since_last_video_ > separate_audio_.packet_threshold &&
packet_feedback.receive_time - last_video_packet_recv_time_ >
separate_audio_.time_threshold) {
active_delay_detector_ = audio_delay_detector_.get();
}
} else {
audio_packets_since_last_video_ = 0;
last_video_packet_recv_time_ =
std::max(last_video_packet_recv_time_, packet_feedback.receive_time);
active_delay_detector_ = video_delay_detector_.get();
}
}
DataSize packet_size = packet_feedback.sent_packet.size;
TimeDelta send_delta = TimeDelta::Zero();
TimeDelta recv_delta = TimeDelta::Zero();
int size_delta = 0;
InterArrivalDelta* inter_arrival_for_packet =
(separate_audio_.enabled && packet_feedback.sent_packet.audio)
? audio_inter_arrival_delta_.get()
: video_inter_arrival_delta_.get();
bool calculated_deltas = inter_arrival_for_packet->ComputeDeltas(
packet_feedback.sent_packet.send_time, packet_feedback.receive_time,
at_time, packet_size.bytes(), &send_delta, &recv_delta, &size_delta);
delay_detector_for_packet->Update(recv_delta.ms<double>(),
send_delta.ms<double>(),
packet_feedback.sent_packet.send_time.ms(),
packet_feedback.receive_time.ms(),
packet_size.bytes(), calculated_deltas);
}
DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
absl::optional<DataRate> link_capacity) {
RateControlInput input(BandwidthUsage::kBwOverusing, link_capacity);
return rate_control_.Update(&input, at_time);
}
DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
absl::optional<NetworkStateEstimate> state_estimate,
bool recovered_from_overuse,
bool in_alr,
Timestamp at_time) {
Result result;
// Currently overusing the bandwidth.
if (active_delay_detector_->State() == BandwidthUsage::kBwOverusing) {
if (acked_bitrate &&
rate_control_.TimeToReduceFurther(at_time, *acked_bitrate)) {
result.updated =
UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
} else if (!acked_bitrate && rate_control_.ValidEstimate() &&
rate_control_.InitialTimeToReduceFurther(at_time)) {
// Overusing before we have a measured acknowledged bitrate. Reduce send
// rate by 50% every 200 ms.
// TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
// so that we (almost) always have a bitrate estimate.
rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, at_time);
result.updated = true;
result.probe = false;
result.target_bitrate = rate_control_.LatestEstimate();
}
} else {
if (probe_bitrate) {
result.probe = true;
result.updated = true;
rate_control_.SetEstimate(*probe_bitrate, at_time);
result.target_bitrate = rate_control_.LatestEstimate();
} else {
result.updated =
UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
result.recovered_from_overuse = recovered_from_overuse;
}
}
BandwidthUsage detector_state = active_delay_detector_->State();
if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
detector_state != prev_state_) {
DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", at_time.ms(), bitrate.bps());
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventBweUpdateDelayBased>(
bitrate.bps(), detector_state));
}
prev_bitrate_ = bitrate;
prev_state_ = detector_state;
}
result.delay_detector_state = detector_state;
return result;
}
bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
absl::optional<DataRate> acked_bitrate,
DataRate* target_rate) {
const RateControlInput input(active_delay_detector_->State(), acked_bitrate);
*target_rate = rate_control_.Update(&input, at_time);
return rate_control_.ValidEstimate();
}
void DelayBasedBwe::OnRttUpdate(TimeDelta avg_rtt) {
rate_control_.SetRtt(avg_rtt);
}
bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
DataRate* bitrate) const {
// Currently accessed from both the process thread (see
// ModuleRtpRtcpImpl::Process()) and the configuration thread (see
// Call::GetStats()). Should in the future only be accessed from a single
// thread.
RTC_DCHECK(ssrcs);
RTC_DCHECK(bitrate);
if (!rate_control_.ValidEstimate())
return false;
*ssrcs = {kFixedSsrc};
*bitrate = rate_control_.LatestEstimate();
return true;
}
void DelayBasedBwe::SetStartBitrate(DataRate start_bitrate) {
RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: "
<< ToString(start_bitrate);
rate_control_.SetStartBitrate(start_bitrate);
}
void DelayBasedBwe::SetMinBitrate(DataRate min_bitrate) {
// Called from both the configuration thread and the network thread. Shouldn't
// be called from the network thread in the future.
rate_control_.SetMinBitrate(min_bitrate);
}
TimeDelta DelayBasedBwe::GetExpectedBwePeriod() const {
return rate_control_.GetExpectedBandwidthPeriod();
}
} // namespace webrtc
|