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gerrit-public.fairphone.software
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platform
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external
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webrtc
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672a48d0d2b96ad6f80558247f34e70a975d1a2b
672a48d
Add sprang@ as owner in modules/rtp_rtcp
by Erik Språng
· 5 years ago
8f319a3
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
by Alessio Bazzica
· 5 years ago
fab3460
Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
by Alessio Bazzica
· 5 years ago
97321b6
Adds test for experimental remote estimate SDP negotiation.
by Sebastian Jansson
· 5 years ago
463d44a
Don't crash when simulcast layer count is different from RID count
by Florent Castelli
· 6 years ago
418f0c5
Fix rtp_analyzer tool
by Alessio Bazzica
· 5 years ago
7db19e0
Report congestion window updates on GoogCC time updates
by Evan Shrubsole
· 5 years ago
a9fbb22
Add a field trial for older applications to reduce the simulcast layer count
by Florent Castelli
· 5 years ago
f640b87
Populate y-axis categorical labels in event log visualizer.
by Bjorn Terelius
· 5 years ago
9973933
Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
by Chen Xing
· 5 years ago
f2d97b8
Add usage message to event_log_visualizer.
by Mirko Bonadei
· 5 years ago
aa59eca
Move RtpPacketSender and merge it with RtpPacketPacer.
by Erik Språng
· 5 years ago
55c4a42
Add congestion window values to GoogCcPrinter
by Evan Shrubsole
· 5 years ago
c4f047d
Make WebRTC-Pacer-LegacyPacketReferencing default off
by Erik Språng
· 6 years ago
414e1b4
Roll chromium_revision 5e244999c5..fdd7489f1f (680219:680383)
by chromium-webrtc-autoroll
· 5 years ago
6cacef2
Reset packet history on ssrc/seqno reset
by Erik Språng
· 5 years ago
a57711c
Fix issue with TransmissionOffset using new pacer code path
by Erik Språng
· 5 years ago
54d9602
Add y-axis tick labels.
by Bjorn Terelius
· 5 years ago
46bbdec
Allow AbsSendTime extension to be used for audio streams.
by Sebastian Jansson
· 5 years ago
e1795f4
Adds remote estimate RTCP packet.
by Sebastian Jansson
· 5 years ago
1796a82
Add performance optimization for empty `RtpPacketInfos`.
by Chen Xing
· 5 years ago
bc6f113
Roll chromium_revision 30f282ecdc..5e244999c5 (680117:680219)
by chromium-webrtc-autoroll
· 5 years ago
f781bb5
[Unit test] Add check to prevent segfault on empty vector.
by Yves Gerey
· 5 years ago
b9f5989
Remove legacy/unused RtpPacketHistory::StorageMode::kStore
by Erik Språng
· 6 years ago
d48fbfc
Roll chromium_revision 4a2e9a72c6..30f282ecdc (680003:680117)
by chromium-webrtc-autoroll
· 5 years ago
66c5bdf
Roll chromium_revision 54f75614d2..4a2e9a72c6 (679885:680003)
by chromium-webrtc-autoroll
· 5 years ago
74a1b4b
Only include payload in bytes sent/received.
by Bjorn A Mellem
· 5 years ago
cfefa0a
Revert "Record audio/video bytes sent in analyzer stream stats."
by Mirko Bonadei
· 5 years ago
61689ab
Make api/video:encoded_image public.
by Mirko Bonadei
· 6 years ago
cbc91efa
Improve low bandwidth audio test instrumentatin, fix PC test
by Artem Titov
· 5 years ago
85ef3ea
Roll chromium_revision 4bcccabeb9..54f75614d2 (679778:679885)
by chromium-webrtc-autoroll
· 6 years ago
2ac9503
Roll chromium_revision 753411f0de..4bcccabeb9 (679672:679778)
by chromium-webrtc-autoroll
· 6 years ago
e32cb72
Roll chromium_revision 868676c219..753411f0de (679562:679672)
by chromium-webrtc-autoroll
· 6 years ago
8fcf354
[tsan] Suppress all of libpulsecommon*.so, following chromium.
by Yves Gerey
· 6 years ago
3f53edb
Add json output arg to mb gen and analyze.
by Debrian Figueroa
· 6 years ago
97273d0
Roll chromium_revision 9f7a1184bf..868676c219 (679448:679562)
by chromium-webrtc-autoroll
· 6 years ago
d978cb4
Record audio/video bytes sent in analyzer stream stats.
by Bjorn A Mellem
· 6 years ago
432fe68
[Cleanup] Remove write-only member _sndCardRecDelay.
by Yves Gerey
· 6 years ago
b93a245
[Unit tests] Remove race condition and dangling pointer to mock.
by Yves Gerey
· 6 years ago
e34d62c
Revert "Makes send side network estimation opt-in."
by Sebastian Jansson
· 6 years ago
f24729b
Add usage message to rtc_event_log2rtp_dump.
by Mirko Bonadei
· 6 years ago
7ea6b29
Adds improved estimate bounded backoff to AimdRateControl.
by Sebastian Jansson
· 6 years ago
c36c8e1
Makes send side network estimation opt-in.
by Sebastian Jansson
· 6 years ago
6982f60
Remove empty OWNERS file.
by Mirko Bonadei
· 6 years ago
39483c6
Migrate some Vp8 simulcast and screen share tests on PC framework
by Artem Titov
· 6 years ago
ec35803
Revert "Don't use all_dependent_configs for sdk frameworks"
by Mirko Bonadei
· 6 years ago
e9ff992
Fix isac_fix_test perf reporting.
by Mirko Bonadei
· 6 years ago
84d5d9f
Roll chromium_revision 38f67470c9..9f7a1184bf (679342:679448)
by chromium-webrtc-autoroll
· 6 years ago
d518f39
Roll chromium_revision 84818c0e32..38f67470c9 (679230:679342)
by chromium-webrtc-autoroll
· 6 years ago
650df38
Roll chromium_revision eec6819794..84818c0e32 (679083:679230)
by chromium-webrtc-autoroll
· 6 years ago
22ff9fc
Removes overuse predictor.
by Sebastian Jansson
· 6 years ago
71c52a8
Roll chromium_revision e02114c8fa..eec6819794 (678980:679083)
by chromium-webrtc-autoroll
· 6 years ago
fefa77c
Add pthread thread-local storage support for ScopedYieldPolicy
by Steve Anton
· 6 years ago
836ab13
Remove rtc::Flag.
by Mirko Bonadei
· 6 years ago
5693c26
Remove rtc_tools/frame_editing.
by Mirko Bonadei
· 6 years ago
ef3eda9
Allow using more jni targets on Linux
by Oleh Prypin
· 6 years ago
1afe657
[Sanitizers] Disable tests at compile-time rather than run-time.
by Yves Gerey
· 6 years ago
21f2fc9
Remove the non-useful rtx payload padding option
by Erik Språng
· 6 years ago
76c89da
Add usage message to peerconnection_server.
by Mirko Bonadei
· 6 years ago
249bade
Add usage message to rgba_to_i420_converter.
by Mirko Bonadei
· 6 years ago
0f6191d
RtpSender::GeneratePadding() fixes for new PacedSender code path
by Erik Språng
· 6 years ago
d70d80d
Add support of negotiating Vp9 SVC in PC test framework.
by Artem Titov
· 6 years ago
7ddca16
Add usage message to reference_less_video_analysis.
by Mirko Bonadei
· 6 years ago
4e72245
Add usage message to psnr_ssim_analyzer.
by Mirko Bonadei
· 6 years ago
06cdb23
Add usage message to rtp_generator.
by Mirko Bonadei
· 6 years ago
857ad62
Remove priority_rate from AudioStreamConfig.
by Jonas Olsson
· 6 years ago
824fb38
Remove anonymous namespace around ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
2ab97f6
Migrate WebRTC test infra to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
63741c7
Don't use all_dependent_configs for sdk frameworks
by Oleh Prypin
· 6 years ago
1a49c13
Roll chromium_revision 52ad041d01..e02114c8fa (678860:678980)
by chromium-webrtc-autoroll
· 6 years ago
43bdcd3
Roll chromium_revision 43851562cb..52ad041d01 (678702:678860)
by chromium-webrtc-autoroll
· 6 years ago
284c302
Roll chromium_revision a87860686e..43851562cb (678573:678702)
by chromium-webrtc-autoroll
· 6 years ago
6704df9
Minor threading-model fix for ADM2 on Windows
by henrika
· 6 years ago
d8c6ec4
Adds support for disabling autostart in ADM2 for Windows
by henrika
· 6 years ago
594597c
Add ability to turn on conference mode during simulcast in PC framework.
by Artem Titov
· 6 years ago
596ed25
Don't assume all simulcast screenshare have 2 temporal layers
by Florent Castelli
· 6 years ago
ee0550c
[Unit tests] Show skipped tests instead of painting them green.
by Yves Gerey
· 6 years ago
79b6980
[PeerConnection] Implement restartIce().
by Henrik Boström
· 6 years ago
b41d5f1
Fix CE in rtp_generator. Add it to default build target
by Ilya Nikolaevskiy
· 6 years ago
40e0d8e
Roll chromium_revision 59bca7b0c1..a87860686e (678465:678573)
by chromium-webrtc-autoroll
· 6 years ago
42343b8
Roll chromium_revision 5f1aeb93dc..59bca7b0c1 (678364:678465)
by chromium-webrtc-autoroll
· 6 years ago
fcf3a87
Android: Expose setting custom visible fraction values for video layout
by Magnus Jedvert
· 6 years ago
b04e256
Roll chromium_revision b5dcfdfc7f..5f1aeb93dc (678230:678364)
by chromium-webrtc-autoroll
· 6 years ago
7f8dbe1
Add config to specify raw audio priority bitrate including overhead.
by Christoffer Rodbro
· 6 years ago
3a51b0e
Reland "Reland "Add wrapper to normalize flags.""
by Oleh Prypin
· 6 years ago
0be40bf
Switch pc client and stunprober to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
f43cc69
Roll chromium_revision 331304131f..b5dcfdfc7f (678125:678230)
by chromium-webrtc-autoroll
· 6 years ago
134aeee
Allow using base jni targets on Linux
by Oleh Prypin
· 6 years ago
0182a03
Reland "Remove the injectable bitrate allocation strategy API."
by Jonas Olsson
· 6 years ago
ea3dddf
Use capacity bounds in AimdRateControl if available.
by Sebastian Jansson
· 6 years ago
17bfafe
Roll chromium_revision eca56ac050..331304131f (677993:678125)
by chromium-webrtc-autoroll
· 6 years ago
b4b52ec
Roll chromium_revision 395aebd4f5..eca56ac050 (677872:677993)
by chromium-webrtc-autoroll
· 6 years ago
b7e8815
Roll chromium_revision 67b5429c0c..395aebd4f5 (677707:677872)
by chromium-webrtc-autoroll
· 6 years ago
162ddb4
Remove +absl/flags exceptions from non root DEPS files.
by Mirko Bonadei
· 6 years ago
08da49d
rtp_test_utils: remove unnecessary dep
by Alessio Bazzica
· 6 years ago
b88fd31
New pacer: keepalive fix, unittests coverage
by Erik Språng
· 6 years ago
bb80c13
Guard against clang-format wrong behavior.
by Yves Gerey
· 6 years ago
12849c7
Revert "Reland "Add wrapper to normalize flags.""
by Mirko Bonadei
· 6 years ago
4b091f4
Switch event_log_visualizer to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
3d9b191
Roll chromium_revision ade23986de..67b5429c0c (676659:677707)
by chromium-webrtc-autoroll
· 6 years ago
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