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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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5932149c9aeaa7679ad0bc3183047766832ca907
5932149
Use different restrictions of acked bitrate lag depending on operating point.
by stefan
· 8 years ago
a790d83
Wire up rtcp xr target bitrate on receive side.
by sprang
· 8 years ago
9afbfc0
Roll chromium_revision f50152dfc4..dd10720676 (435897:435924)
by buildbot
· 8 years ago
0f01c7f
Added tool for reference less video analysis (go/refless-video-analysis)
by charujain
· 8 years ago
b2250e5
New gn target video_frame_api.
by nisse
· 8 years ago
969b12f
Remove xdisplaycheck
by kjellander
· 8 years ago
df28e47
fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther.
by howtofly
· 8 years ago
a28a1b9
Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan
by henrik.lundin
· 8 years ago
5c71166
VP8DecoderImpl: Fix uninitialized memory crash
by magjed
· 8 years ago
00bceb1
Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
by ossu
· 8 years ago
e066b30
Remove API-related #defines from voice_engine_configurations.h
by henrik.lundin
· 8 years ago
15cf9f9
Roll chromium_revision 601e4f48a3..f50152dfc4 (435870:435897)
by buildbot
· 8 years ago
b336392
Sanity check parsed QP values from H264 bitstream
by kthelgason
· 8 years ago
864f58b
Roll chromium_revision 353f713f3d..601e4f48a3 (435846:435870)
by buildbot
· 8 years ago
ea3d923
Roll chromium_revision c6d437f401..353f713f3d (435797:435846)
by buildbot
· 8 years ago
51cb31c
Roll chromium_revision 47f73d2d11..c6d437f401 (435727:435797)
by buildbot
· 8 years ago
b465980
In end-to-end PeerConnection tests, allow video to be downscaled.
by deadbeef
· 8 years ago
897530e
Roll chromium_revision bffc40ae72..47f73d2d11 (435660:435727)
by buildbot
· 8 years ago
8f89bff
Revert of Disabled flaky P2PTestConductor tests on ASAN and MSAN. (patchset #1 id:1 of https://codereview.webrtc.org/2539103002/ )
by deadbeef
· 8 years ago
c6b6e09
Relaxing timeouts for TestMediaMonitor.
by deadbeef
· 8 years ago
8f425f9
Relaxing DCHECK for packets sent before SRTP is enabled.
by deadbeef
· 8 years ago
183d51a
Roll chromium_revision 3045382d44..bffc40ae72 (435618:435660)
by buildbot
· 8 years ago
f29e05d
Add linearly spaced counting histograms
by henrik.lundin
· 8 years ago
1454669
Cleanup RtpHeaderExtensionMap removing use of two legacy functions
by danilchap
· 8 years ago
73f2ee2
Roll chromium_revision a6a331af3e..3045382d44 (435597:435618)
by buildbot
· 8 years ago
182e4a4
Remove bitrate cap for AdaptiveVideoSource and increase other caps to 25 Mbps.
by terelius
· 8 years ago
2d9c877
Revert of Whitespace.
by tandrii
· 8 years ago
106edae
Revert of Whitespace.
by Andrii Shyshkalov
· 8 years ago
aa7a9a2
Whitespace.
by Andrii Shyshkalov
· 8 years ago
3b56e1f
Whitespace.
by tandrii
· 8 years ago
1a646ee
Wire up BitrateAllocation to be sent as RTCP TargetBitrate
by sprang
· 8 years ago
5e38c96
Wire up RTCP XR target bitrate in rtp/rtcp module
by sprang
· 8 years ago
b57e0f6
Roll chromium_revision 698c4714e8..a6a331af3e (435589:435597)
by buildbot
· 8 years ago
5e13d41
Remove limit on how often quality scaling downscales
by kthelgason
· 8 years ago
86cf9a2
Increase test timeout to combat flakiness.
by kthelgason
· 8 years ago
e90adce
Remove OnLocalSsrcChanged
by mflodman
· 8 years ago
847f294
Roll chromium_revision b5ce7e0a17..698c4714e8 (435129:435589)
by buildbot
· 8 years ago
665bc3c
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
by magjed
· 8 years ago
2bc324c
Add method on AVFoundation capturer to adapt output format.
by kthelgason
· 8 years ago
dd40702
Move VideoDecoder::Create() logic to separate internal video decoder factory
by magjed
· 8 years ago
aa354c9
Rename full_stack.cc to full_stack_tests.cc.
by brandtr
· 8 years ago
a974d76
Enable VideoToolbox encoder on mac
by kthelgason
· 8 years ago
024c90e
Whitespace CL to trigger bots.
by Henrik Kjellander
· 8 years ago
3a70cc3
Whitespace change to trigger bots.
by Henrik Kjellander
· 8 years ago
5c13c33
Roll chromium_revision b66d8ae9dc..b5ce7e0a17 (435081:435129)
by buildbot
· 8 years ago
ed6e077
Make SurfaceTextureHelper and I420Frame public in Java.
by skvlad
· 8 years ago
a15948c
Change assert to RTC_DCHECK in bwe_test_logging.cc
by terelius
· 8 years ago
3a9bc17
Allow custom drawers to be added to framelisteners.
by sakal
· 8 years ago
06c1d64
Prep to remove API-related #defines from voice_engine_configurations.h
by henrik.lundin
· 8 years ago
9332b7d
Reland "Update rtt on audio only calls".
by michaelt
· 8 years ago
93c5d03
Start gathering perf data for VP8 + FlexFEC.
by brandtr
· 8 years ago
13d38fb
Delete all of the video_processing module but the denoiser code.
by nisse
· 8 years ago
13f1a0a
Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel.
by stefan
· 8 years ago
127387e
Delete nalu parser in mediaencoder
by kthelgason
· 8 years ago
2305b70
Remove unused build override
by kthelgason
· 8 years ago
cb861e0
Templatize percentile_filter.h and move it to base/analytics.
by terelius
· 8 years ago
4b9a2cb
Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."
by minyue
· 8 years ago
26bddb9
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
by ehmaldonado
· 8 years ago
18f7c8d
Remove warning suppression from VideoToolboxEncoder
by kthelgason
· 8 years ago
076c011
Change unit of logged bitrate stats in bytes/s to bits/s.
by asapersson
· 8 years ago
aff9636
Greatly reduce number of level controller tests.
by phoglund
· 8 years ago
6a2e20a
Make sure GetLastError on a PlatformThread return an error that is relevant to the thread.
by perkj
· 8 years ago
b49fc14
RtpDataEngine, FindCodecByName: Don't reassign codecs
by magjed
· 8 years ago
998df1d
Create webrtc/sdk/android folder
by magjed
· 8 years ago
78b4d56
Relanding "Pass time constant to bwe smoothing filter."
by minyue
· 8 years ago
8d66a5a
Disabled flaky P2PTestConductor tests on ASAN and MSAN.
by ossu
· 8 years ago
0245da0
Move ownership of PacketRouter from CongestionController to Call.
by nisse
· 8 years ago
06251f1
Reduce ProbeController::kDefaultMaxProbingBitrateBps to 10 mbps.
by philipel
· 8 years ago
f4a5942
Disabled all ScreenCapturerIntegrationTests on Windows
by ossu
· 8 years ago
3a864d2
MB: Add swarming bots in the FYI waterfall and remove memcheck swarming.
by ehmaldonado
· 8 years ago
b7e1dd7
Revert of Adding memcheck suppression. (patchset #1 id:1 of https://codereview.webrtc.org/2537563003/ )
by philipel
· 8 years ago
706a45e
Added missing include to fix waterfall compile error.
by hbos
· 8 years ago
f15a2c5
Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom.
by nisse
· 8 years ago
0583b28
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
by hbos
· 8 years ago
0c43f77
Update video histograms that do not have a minimum lifetime limit before being recorded.
by asapersson
· 8 years ago
759e0b7
Fix memory leak in video_coding::PacketBuffer::InsertPacket.
by philipel
· 8 years ago
be74270
Calculate JitterBufferDelayInMs in the new jitter buffer.
by philipel
· 8 years ago
e69b468
Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
by minyue
· 8 years ago
1731c9c
Use swap instead of copy in RtcHistogram::GetAndReset.
by asapersson
· 8 years ago
84e56d5
Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
by minyue
· 8 years ago
097529f
Remove 3 defines in voice_engine_configurations.h
by henrik.lundin
· 8 years ago
e61fbff
Use RotateDesktopFrame in DirectX capturer
by zijiehe
· 8 years ago
166e59a
Enable ScreenCapturerIntegrationTests
by zijiehe
· 8 years ago
6a46cf7
Roll chromium_revision db14e1adbd..b66d8ae9dc (435041:435081)
by buildbot
· 8 years ago
c9e80ee
Adding packet overhead to audio network adaptor.
by minyue
· 8 years ago
821dc7a
Roll chromium_revision 683745f53c..db14e1adbd (434997:435041)
by buildbot
· 8 years ago
a332877
Remove overhead from video bitrate.
by michaelt
· 8 years ago
c4dc4a5
Adding RTCStatsIntegrationTest to memcheck supressions.
by deadbeef
· 8 years ago
75f9d8c
Roll chromium_revision ffe8e7b51d..683745f53c (434954:434997)
by buildbot
· 8 years ago
290d43a
Add a new UMA metric in APM to track incoming capture-side audio level
by henrik.lundin
· 8 years ago
939e08f
Added webrtc/audio/utility directory and empty GN target.
by aleloi
· 8 years ago
ee414d9
Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error.
by philipel
· 8 years ago
ad6f646
Use //build/dotfile_settings.gni to reduce blocked auto-rolls
by kjellander
· 8 years ago
768d625
Fix spelling mistake in RTP module declaration.
by brandtr
· 8 years ago
b890c95c
Replace some asserts with DCHECKs
by kwiberg
· 8 years ago
5049942
Refactor RMSLevel and give it new functionality
by henrik.lundin
· 8 years ago
1308c69
Roll chromium_revision 0496be2799..ffe8e7b51d (434847:434954)
by buildbot
· 8 years ago
f17cae2
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
by kjellander
· 8 years ago
668eb3b
Add overhead to transport feedback observer.
by michaelt
· 8 years ago
19223ac
Ignore newly added resource files.
by charujain
· 8 years ago
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