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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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106edaeb854e68c930960d2a6d899fa0925240ab
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
455b512
Landmine to clobber Windows builders
by Henrik Kjellander
· 8 years ago
1b5b22d
Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
by charujain
· 8 years ago
db346a7
RTCStatsIntegrationTest added.
by hbos
· 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
65e5f5a
Roll chromium_revision d74a300097..0496be2799 (434704:434847)
by buildbot
· 8 years ago
bdd6f4c
Adding memcheck suppression.
by deadbeef
· 8 years ago
6cf94a0
Only use BoringSSL time callback in unit tests.
by deadbeef
· 8 years ago
352444f
RTC_[D]CHECK_op: Remove superfluous casts
by kwiberg
· 8 years ago
af476c7
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
by kwiberg
· 8 years ago
80ed35e
Implement periodic bandwidth probing in application-limited region.
by sergeyu
· 8 years ago
bf22be9
Roll chromium_revision 2b5aa49038..d74a300097 (434640:434704)
by buildbot
· 8 years ago
fd87f4a
Opus: Move complexity variable out of conditional build flag
by henrik.lundin
· 8 years ago
1bc3146
Disable more VideoProcessorIntegrationTest tests on Linux 32-bit
by Henrik Kjellander
· 8 years ago
bb58435
Fix potential synchronization issues with framelisteners in EglRenderer.
by sakal
· 8 years ago
266f0a4
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
by philipel
· 8 years ago
d1aaaaa
Remove surface size mismatch logic from EglRenderer.
by sakal
· 8 years ago
6287e82
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
by ossu
· 8 years ago
7703b27
Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots.
by philipel
· 8 years ago
ceecea4
Pass selected cricket::VideoCodec down to internal H264 encoder
by magjed
· 8 years ago
20dce34
Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called.
by philipel
· 8 years ago
e1a13f8
MB: Remove a --target-devices-file flag for JUnit tests on android.
by ehmaldonado
· 8 years ago
a8eb756
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 8 years ago
9abbf5a
Pass time constanct to bwe smoothing filter.
by michaelt
· 8 years ago
ffc6118
Don't cache video codec list in VideoEngine2.
by brandtr
· 8 years ago
ec1a670
Only create |remote_rate| when needed in RemoteBitrateEstimatorSingleStream.
by Rasmus Brandt
· 8 years ago
fb4a37a
Add memcheck suppressions.
by ehmaldonado
· 8 years ago
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