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Side by Side Diff: webrtc/video/receive_statistics_proxy.cc

Issue 1616153005: Switch to use new implementation in metrics.h. (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) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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
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30 stats_.ssrc = ssrc; 30 stats_.ssrc = ssrc;
31 } 31 }
32 32
33 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() { 33 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
34 UpdateHistograms(); 34 UpdateHistograms();
35 } 35 }
36 36
37 void ReceiveStatisticsProxy::UpdateHistograms() { 37 void ReceiveStatisticsProxy::UpdateHistograms() {
38 int fraction_lost = report_block_stats_.FractionLostInPercent(); 38 int fraction_lost = report_block_stats_.FractionLostInPercent();
39 if (fraction_lost != -1) { 39 if (fraction_lost != -1) {
40 RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.ReceivedPacketsLostInPercent", 40 RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
41 fraction_lost); 41 fraction_lost);
42 } 42 }
43 const int kMinRequiredSamples = 200; 43 const int kMinRequiredSamples = 200;
44 int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount()); 44 int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
45 if (samples > kMinRequiredSamples) { 45 if (samples > kMinRequiredSamples) {
46 RTC_HISTOGRAM_COUNTS_SPARSE_100("WebRTC.Video.RenderFramesPerSecond", 46 RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
47 round(render_fps_tracker_.ComputeTotalRate())); 47 round(render_fps_tracker_.ComputeTotalRate()));
48 RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Video.RenderSqrtPixelsPerSecond", 48 RTC_HISTOGRAM_COUNTS_100000(
49 "WebRTC.Video.RenderSqrtPixelsPerSecond",
49 round(render_pixel_tracker_.ComputeTotalRate())); 50 round(render_pixel_tracker_.ComputeTotalRate()));
50 } 51 }
51 int width = render_width_counter_.Avg(kMinRequiredSamples); 52 int width = render_width_counter_.Avg(kMinRequiredSamples);
52 int height = render_height_counter_.Avg(kMinRequiredSamples); 53 int height = render_height_counter_.Avg(kMinRequiredSamples);
53 if (width != -1) { 54 if (width != -1) {
54 RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.ReceivedWidthInPixels", 55 RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
55 width); 56 RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
56 RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.ReceivedHeightInPixels",
57 height);
58 } 57 }
59 int qp = qp_counters_.vp8.Avg(kMinRequiredSamples); 58 int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
60 if (qp != -1) 59 if (qp != -1)
61 RTC_HISTOGRAM_COUNTS_SPARSE_200("WebRTC.Video.Decoded.Vp8.Qp", qp); 60 RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
62 61
63 // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and 62 // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
64 // not per frame. Change decode time to include every frame. 63 // not per frame. Change decode time to include every frame.
65 const int kMinRequiredDecodeSamples = 5; 64 const int kMinRequiredDecodeSamples = 5;
66 int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples); 65 int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
67 if (decode_ms != -1) 66 if (decode_ms != -1)
68 RTC_HISTOGRAM_COUNTS_SPARSE_1000("WebRTC.Video.DecodeTimeInMs", decode_ms); 67 RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
69 68
70 int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples); 69 int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
71 if (delay_ms != -1) 70 if (delay_ms != -1)
72 RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.OnewayDelayInMs", delay_ms); 71 RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
73 } 72 }
74 73
75 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const { 74 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
76 rtc::CritScope lock(&crit_); 75 rtc::CritScope lock(&crit_);
77 return stats_; 76 return stats_;
78 } 77 }
79 78
80 void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) { 79 void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
81 rtc::CritScope lock(&crit_); 80 rtc::CritScope lock(&crit_);
82 stats_.current_payload_type = payload_type; 81 stats_.current_payload_type = payload_type;
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208 ++num_samples; 207 ++num_samples;
209 } 208 }
210 209
211 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const { 210 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
212 if (num_samples < min_required_samples || num_samples == 0) 211 if (num_samples < min_required_samples || num_samples == 0)
213 return -1; 212 return -1;
214 return sum / num_samples; 213 return sum / num_samples;
215 } 214 }
216 215
217 } // namespace webrtc 216 } // namespace webrtc
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