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
|
/*
* Copyright (c) 2013 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 "webrtc/video/receive_statistics_proxy.h"
#include <cmath>
#include "webrtc/base/checks.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"
namespace webrtc {
ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
: clock_(clock),
// 1000ms window, scale 1000 for ms to s.
decode_fps_estimator_(1000, 1000),
renders_fps_estimator_(1000, 1000),
render_fps_tracker_(100u, 10u),
render_pixel_tracker_(100u, 10u) {
stats_.ssrc = ssrc;
}
ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
UpdateHistograms();
}
void ReceiveStatisticsProxy::UpdateHistograms() {
int fraction_lost = report_block_stats_.FractionLostInPercent();
if (fraction_lost != -1) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
fraction_lost);
}
const int kMinRequiredSamples = 200;
int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
if (samples > kMinRequiredSamples) {
RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
static_cast<int>(render_fps_tracker_.ComputeTotalRate()));
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.RenderSqrtPixelsPerSecond",
static_cast<int>(render_pixel_tracker_.ComputeTotalRate()));
}
int width = render_width_counter_.Avg(kMinRequiredSamples);
int height = render_height_counter_.Avg(kMinRequiredSamples);
if (width != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
}
int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
if (qp != -1)
RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
// TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
// not per frame. Change decode time to include every frame.
const int kMinRequiredDecodeSamples = 5;
int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
if (decode_ms != -1)
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
if (delay_ms != -1)
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
}
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
rtc::CritScope lock(&crit_);
return stats_;
}
void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
rtc::CritScope lock(&crit_);
stats_.current_payload_type = payload_type;
}
void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
unsigned int bitrate_bps) {
rtc::CritScope lock(&crit_);
stats_.network_frame_rate = framerate;
stats_.total_bitrate_bps = bitrate_bps;
}
void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
int jitter_buffer_ms,
int min_playout_delay_ms,
int render_delay_ms,
int64_t rtt_ms) {
rtc::CritScope lock(&crit_);
stats_.decode_ms = decode_ms;
stats_.max_decode_ms = max_decode_ms;
stats_.current_delay_ms = current_delay_ms;
stats_.target_delay_ms = target_delay_ms;
stats_.jitter_buffer_ms = jitter_buffer_ms;
stats_.min_playout_delay_ms = min_playout_delay_ms;
stats_.render_delay_ms = render_delay_ms;
decode_time_counter_.Add(decode_ms);
// Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
// render delay).
delay_counter_.Add(target_delay_ms + rtt_ms / 2);
}
void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) {
rtc::CritScope lock(&crit_);
if (stats_.ssrc != ssrc)
return;
stats_.rtcp_packet_type_counts = packet_counter;
}
void ReceiveStatisticsProxy::StatisticsUpdated(
const webrtc::RtcpStatistics& statistics,
uint32_t ssrc) {
rtc::CritScope lock(&crit_);
// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
// receive stats from one of them.
if (stats_.ssrc != ssrc)
return;
stats_.rtcp_stats = statistics;
report_block_stats_.Store(statistics, ssrc, 0);
}
void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
rtc::CritScope lock(&crit_);
// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
// receive stats from one of them.
if (stats_.ssrc != ssrc)
return;
stats_.c_name = cname;
}
void ReceiveStatisticsProxy::DataCountersUpdated(
const webrtc::StreamDataCounters& counters,
uint32_t ssrc) {
rtc::CritScope lock(&crit_);
if (stats_.ssrc != ssrc)
return;
stats_.rtp_stats = counters;
}
void ReceiveStatisticsProxy::OnDecodedFrame() {
uint64_t now = clock_->TimeInMilliseconds();
rtc::CritScope lock(&crit_);
decode_fps_estimator_.Update(1, now);
stats_.decode_frame_rate = decode_fps_estimator_.Rate(now);
}
void ReceiveStatisticsProxy::OnRenderedFrame(int width, int height) {
RTC_DCHECK_GT(width, 0);
RTC_DCHECK_GT(height, 0);
uint64_t now = clock_->TimeInMilliseconds();
rtc::CritScope lock(&crit_);
renders_fps_estimator_.Update(1, now);
stats_.render_frame_rate = renders_fps_estimator_.Rate(now);
render_width_counter_.Add(width);
render_height_counter_.Add(height);
render_fps_tracker_.AddSamples(1);
render_pixel_tracker_.AddSamples(sqrt(width * height));
}
void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
uint32_t frameRate) {
}
void ReceiveStatisticsProxy::OnFrameCountsUpdated(
const FrameCounts& frame_counts) {
rtc::CritScope lock(&crit_);
stats_.frame_counts = frame_counts;
}
void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
rtc::CritScope lock(&crit_);
stats_.discarded_packets = discarded_packets;
}
void ReceiveStatisticsProxy::OnPreDecode(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info) {
if (codec_specific_info == nullptr || encoded_image.qp_ == -1) {
return;
}
if (codec_specific_info->codecType == kVideoCodecVP8) {
qp_counters_.vp8.Add(encoded_image.qp_);
}
}
void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
sum += sample;
++num_samples;
}
int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
if (num_samples < min_required_samples || num_samples == 0)
return -1;
return sum / num_samples;
}
} // namespace webrtc
|
