Log - 6c373cccbb7bcf052431fb9f80cf2c13c1af9933 - platform/external/webrtc

e2fd86a Move encoder metadata into EncoderInfo struct. by Erik Språng · 6 years ago
17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
bb081a6 Update sending resolution only on top spatial layer frame. by Sergey Silkin · 6 years ago
d3b8c63 Reland "Add spatial index to EncodedImage." by Niels Möller · 6 years ago
5a998d7 Revert "Add spatial index to EncodedImage." by Niels Moller · 6 years ago
da0898d Add spatial index to EncodedImage. by Niels Möller · 6 years ago
2377588 Add accessor methods for RTP timestamp of EncodedImage. by Niels Möller · 6 years ago
20317f9 Clear encoded frame map on gap in timestamp. by Åsa Persson · 6 years ago
213618e New api function CreateVideoStreamEncoder. by Niels Möller · 6 years ago
43800f9 Generalize SimulcastEncoderAdapter, use for H264 & VP8. by Sergio Garcia Murillo · 6 years ago
6f440ed Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." by Mirko Bonadei · 6 years ago
07efe43 Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. by Sergio Garcia Murillo · 6 years ago
b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
97e0488 Delete unused stats for preferred_bitrate. by Niels Möller · 6 years ago
875841d Exclude initial adapt downs in stats for quality adapt changes per minute. by Åsa Persson · 7 years ago
8e07c13 Optional: Use nullopt and implicit construction in /video by Oskar Sundbom · 7 years ago
aa329e7 Reland: googBandwidthLimitedResolution stat is not always set depending on configuration. by Åsa Persson · 7 years ago
62e9ebe Revert "googBandwidthLimitedResolution stat is not always set depending on configuration." by Guido Urdaneta · 7 years ago
59283e4 googBandwidthLimitedResolution stat is not always set depending on configuration. by Åsa Persson · 7 years ago
c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
45bbc8a Change forced software encoder fallback for VP8 to be only based on resolution and not bitrate. by Åsa Persson · 7 years ago
0122e84 Reland "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
ca0ed63 Revert "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
af721b7 Remove sent framerate and bitrate calculations from MediaOptimization. by Åsa Persson · 7 years ago
8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 7 years ago
92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/send_statistics_proxy_unittest.cc]
186d9c3 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
cc3d442 Rename ViEEncoder to VideoStreamEncoder by mflodman · 7 years ago
09f0561 Update adaptation stats to support degradations in both resolution and framerate. by asapersson · 7 years ago
f4e44af Do not report cpu limited resolution stats when degradation preference is disabled and no scaling is done. by asapersson · 8 years ago
0944a80 Update stats for cpu/quality adaptation changes to excluded time when video is suspended. by asapersson · 8 years ago
6eca98b Add histogram stats for number of cpu/quality adapt changes per minute for sent video streams: by asapersson · 8 years ago
fab6707 Add number of quality adapt changes to VideoSendStream stats. by asapersson · 8 years ago
36e9eb4 Do not report quality limited resolution stats when quality scaler is disabled. by asapersson · 8 years ago
93e1e23 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
3d200bd Remove FlexfecConfig and replace with specific struct in VideoSendStream. by brandtr · 8 years ago
66d4b37 Move histogram for number of pause events to per stream: by asapersson · 8 years ago
0cd27ba Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ ) by kthelgason · 8 years ago
fe04bd4 Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ ) by kthelgason · 8 years ago
0c8c538 Properly report number of quality downscales in stats. by kthelgason · 8 years ago
876222f Move usage of QualityScaler to ViEEncoder. by kthelgason · 8 years ago
320e45a Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
a6a699a Sent bitrate stats are incorrect if FlexFEC is configured: by asapersson · 8 years ago
827cab3 Add qp counter for H264 in SendStatisticsProxy. by asapersson · 8 years ago
803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
87da404 Implement qpSum stat for video send ssrc stats. by sakal · 8 years ago
4ee7046 Add unit tests for bandwidth limited resolution stats in SendStatisticsProxy. by asapersson · 8 years ago
43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
a48ddb7 Add VideoSendStream::Stats::prefered_media_bitrate_bps by Per · 8 years ago
77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
275afc5 Add codec name to CodecSpecificInfo and get the codec name stats from there instead. by perkj · 8 years ago
2e5cfcd Add periodic logging of video stats. by asapersson · 8 years ago
4374a09 Only update codec type histogram if lifetime is long enough (10 sec). by asapersson · 8 years ago
cd349d9 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ ) by sprang · 8 years ago
a49f110 Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ ) by aluebs · 8 years ago
05ce4ae Reland Issue 2061423003: Refactor NACK bitrate allocation by Erik Språng · 8 years ago
e5dd441 Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ ) by sprang · 8 years ago
5fc59e8 Refactor NACK bitrate allocation by Erik Språng · 8 years ago
01d70a3 Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h. by asapersson · 8 years ago
35151f3 Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. by asapersson · 9 years ago
376b192 Remove VideoCodingModule::VCMPacketizationCallback by perkj · 9 years ago
d98d457 Remove "This file includes unit tests" comments. by Peter Boström · 9 years ago
02b3d27 Reland of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #1 id:1 of https://codereview.webrtc.org/1903193002/ ) by kjellander · 9 years ago
a261e61 Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ ) by kjellander · 9 years ago
f5d55aa Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. by perkj · 9 years ago
5265fed Add histogram stats for average QP per frame for VP9 (for sent video streams): by asapersson · 9 years ago
118ef00 Add histogram stats for average QP per frame for VP8 (for sent video streams): by asapersson · 9 years ago
27f982b Replace scoped_ptr with unique_ptr in webrtc/video/ by kwiberg · 9 years ago
22c2b48 Move RTP stats histograms from VieChannel to SendStatisticsProxy. by Erik Språng · 9 years ago
07fb9be Move RTCP histograms from vie_channel to video channel stats proxies. by sprang · 9 years ago
e449915 Measure encoding time on encode callbacks. by Peter Boström · 9 years ago
59bac1a Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest. by asapersson · 9 years ago
1aa420b Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead. by asapersson · 9 years ago
b4a1ae5 Add separate send-side UMA stats for screenshare and video. by sprang · 9 years ago
7083e11 Remove callback_cs_ in ViEEncoder. by Peter Boström · 9 years ago
4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
20f3f94 Clear bitrate stats for unused SSRCs. by Peter Boström · 9 years ago
af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago
d7452a0 Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame." by magjed@webrtc.org · 10 years ago
bcead30 Make the entry point for VideoFrames to webrtc const ref I420VideoFrame. by perkj@webrtc.org · 10 years ago
00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
09c77b9 Add decoder-timing stats to VideoReceiveStream. by pbos@webrtc.org · 10 years ago
49096de DCHECK send DataCountersUpdated for valid SSRCs. by pbos@webrtc.org · 10 years ago
cfd82df Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
273a414 Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
58e2d26 Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). by stefan@webrtc.org · 10 years ago
168f23f Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
4ef438e Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
b10363f Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
be39470 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
1598b80 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
0931570 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
ccd4284 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago