Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
1e0cfd9a462185ba5c434114cf9dcaf160cd68e3
1e0cfd9
Add VP8 and H264 depacketizer fuzzers.
by Peter Boström
· 9 years ago
9d98f21
Roll chromium_revision 68898fb..ddfc1fe (365698:365801)
by kjellander
· 9 years ago
a689b44
Add tracing to NetEqImpl::InsertPacket
by henrik.lundin
· 9 years ago
0eb15ed
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
by kwiberg
· 9 years ago
e376f0f
Add Windows Clang trybots to the default set.
by kjellander
· 9 years ago
e40dedb
Roll chromium_revision 004c7b4..68898fb (365580:365698)
by kjellander
· 9 years ago
a089257
Cleanup use of "do { ... } while (0)".
by torbjorng
· 9 years ago
a54a080
Add ufrag to the ICE candidate signaling.
by honghaiz
· 9 years ago
3514cbe
Add DrFuzz support to webrtc fuzzers.
by pbos
· 9 years ago
7cae30c
Disable warnings failing when using Clang on Windows.
by kjellander
· 9 years ago
9f58795
Roll chromium_revision 2c8eb1f..004c7b4 (365513:365580)
by kjellander
· 9 years ago
361888c
OWNERS: Add * to .gyp{i,} everywhere.
by kjellander@webrtc.org
· 9 years ago
2f29d70
Roll gtest-parallel.
by pbos
· 9 years ago
0bc176b
Further refactored the echo suppressor code:
by peah
· 9 years ago
c482eb3
Don't account for audio in the pacer budget.
by Stefan Holmer
· 9 years ago
5f026d0
Update NetEq network statistics in neteq_unittest.
by minyue
· 9 years ago
4430763
AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders
by kwiberg
· 9 years ago
99b1a32
Retyped the frequency estimate of the comfort noise for the higher band to harmonize the AEC code.
by peah
· 9 years ago
426ae9d
Roll chromium_revision 6e5b8cb..2c8eb1f (365419:365513)
by kjellander
· 9 years ago
a6db495
Move Rent-A-Codec out of CodecManager
by kwiberg
· 9 years ago
a29386c
Make VoiceDetection not a ProcessingComponent (bit exact).
by solenberg
· 9 years ago
672aba3
Fix error prone code in VideoCapturerAndroid
by perkj
· 9 years ago
66085be
Bugfix that fixes the error where the audio processing module is called
by peah
· 9 years ago
54999d4
rtcp::Dlrr block moved into own file and got Parse function
by danilchap
· 9 years ago
29e2f93
Fix NoiseSuppression initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1507683006/.
by solenberg
· 9 years ago
45fd9fe
New macro: RTC_DEPRECATED (for annotating deprecated functions)
by kwiberg
· 9 years ago
ed644d8
Roll chromium_revision bff4606..6e5b8cb (365226:365419)
by kjellander
· 9 years ago
eb45981
Restoring behavior where PeerConnection tracks changes to MediaStreams.
by deadbeef
· 9 years ago
44f0819
Fixing bug where "mid" wasn't preserved across re-offers.
by deadbeef
· 9 years ago
c1316a1
Fix HPF initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1490333004/.
by solenberg
· 9 years ago
95d9851
Add speech encoder to the encoder stack specification struct
by kwiberg
· 9 years ago
7eb914d
Fix incorrect comment
by kwiberg
· 9 years ago
78315b9
Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ )
by Peter Boström
· 9 years ago
f9945b2
Only try to pair protocol matching candidates for creating connections.
by Honghai Zhang
· 9 years ago
949028f
Make LevelEstimation not a ProcessingComponent.
by solenberg
· 9 years ago
5e0218c
Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ )
by tommi
· 9 years ago
5125433
Android: Refactor renderers to allow apps to inject custom shaders
by Magnus Jedvert
· 9 years ago
91941ae
rtcp::VoipMetric block moved into own file and got Parse function
by danilchap
· 9 years ago
32d989b
Disable transport sequence numbers for audio.
by Stefan Holmer
· 9 years ago
10aea22
Roll chromium_revision 53970fd..bff4606 (365141:365226)
by kjellander
· 9 years ago
377b5e6
enabled cpplint for the webrtc/modules/rtp_rtcp directory
by danilchap
· 9 years ago
6eca7e3
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
by tommi
· 9 years ago
6db6cdc
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
by danilchap
· 9 years ago
9638143
Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ )
by perkj
· 9 years ago
e005cf2
[rtp_rtcp] SSRCDatabase class cleaned (including all lint errors)
by danilchap
· 9 years ago
5ea3da2
Base webrtc fuzzers on a template.
by Peter Boström
· 9 years ago
8f09f17
Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script.
by terelius
· 9 years ago
498ae00
Disable ThreadTest.ThreeThreadsInvoke on DrMemory bots.
by Stefan Holmer
· 9 years ago
47a740b
[rtp_rtcp] lint errors about rand() usage fixed.
by danilchap
· 9 years ago
2d36b92
Roll chromium_revision 10bf0e1..53970fd (365000:365141)
by kjellander
· 9 years ago
1588793
Fixing flaky LocalP2PTestSctpDataChannel test.
by deadbeef
· 9 years ago
c9be007
Fixing and re-enabling some flaky PeerConnection tests.
by deadbeef
· 9 years ago
bd29246
Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ )
by deadbeef
· 9 years ago
82ccfcf
Remove unused and rarely used LOG_ macros.
by solenberg
· 9 years ago
e22e1cb
Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ )
by perkj
· 9 years ago
40f349f
[rtp_rtcp] Lint errors cleared from rtp_rtcp/test
by danilchap
· 9 years ago
3207916
Made EglBase an abstract class and cleaned up.
by perkj
· 9 years ago
03960d9
Roll chromium_revision 4bc4277..10bf0e1 (364953:365000)
by kjellander
· 9 years ago
bc14164
Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
by stefan
· 9 years ago
b2f80e3
rtp_rtcp/test/BWEStandAlone deleted as obsolete
by danilchap
· 9 years ago
a78c021
Add APK targets to build libjingle_peerconnection_unittests for Android.
by perkj
· 9 years ago
17821db
Wire up bandwidth limitation info to GetStats and adapt_reason.
by asapersson
· 9 years ago
ac921d7
Add "x"s in the end of a stripped IPv6 address string.
by henrikg
· 9 years ago
38bb8ad
Add test for verifying configured key frame interval for VP9.
by asapersson
· 9 years ago
e5ae6f8
Correcting the check for the return code produced by
by peah
· 9 years ago
1d5c19d
Address comments from code review 1505253004
by tommi
· 9 years ago
4759bfb
Roll chromium_revision 7de03ed..4bc4277 (364770:364953)
by kjellander
· 9 years ago
aa32c3e
Update API for Objective-C RTCIceServer
by hjon
· 9 years ago
cb95f54e
Remove pointless move() to fix build on clang/win.
by Tommi
· 9 years ago
66679dc
Update WARN_UNUSED_RESULT macro to match Chromium's version.
by tfarina
· 9 years ago
be26c07
Roll gtest-parallel.
by pbos
· 9 years ago
b798f38
Roll chromium_revision 710285b..7de03ed (364599:364770)
by kjellander
· 9 years ago
f888bb5
Support for unmixed remote audio into tracks.
by Tommi
· 9 years ago
f67c548
Handle Turn error response to RefreshRequest, CreatePermissionRequest, and ChanelBindRequest
by Honghai Zhang
· 9 years ago
04e9146
Discard old-generation candidates when ICE restarts
by Honghai Zhang
· 9 years ago
43e4e23
Remove thread-id wraparounds in event tracing.
by Peter Boström
· 9 years ago
822bdf9
Remove cricket::VideoEncoderConfig.
by Peter Boström
· 9 years ago
4c1093b
Add FEC producer fuzzing and a unittest for one of the issues found.
by Stefan Holmer
· 9 years ago
5b659c0
Special-case android-arm64 in codec bitexactness tests
by kwiberg
· 9 years ago
b562c33
Remove ancient VoE suppressions.
by solenberg
· 9 years ago
cb23c0d
Adding Opus to RTPencode.
by minyue
· 9 years ago
71f5a9a
This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers.
by Per
· 9 years ago
0b0a88b
Add aecdump support to AppRTCDemo
by aluebs
· 9 years ago
4dfe332
Roll chromium_revision 026b937..710285b (364421:364599)
by kjellander
· 9 years ago
55bcf0f
Fix -Wformat error in Win-Clang build (take 2)
by hans
· 9 years ago
013e83b
Fix -Wformat error in Win-Clang build
by Niklas Enbom
· 9 years ago
cf846ad
Adding stub files needed for https://codereview.webrtc.org/1507973003/
by Taylor Brandstetter
· 9 years ago
7c73bdb
Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor.
by deadbeef
· 9 years ago
ed83edc
Roll chromium_revision 2e451bf..026b937 (364330:364421)
by kjellander
· 9 years ago
6a6f089
in rtp_rtcp module:
by danilchap
· 9 years ago
a1f567a
Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ )
by deadbeef
· 9 years ago
61a90f9
clang/win: Fix -Wextra warnings in webrtc.
by thakis
· 9 years ago
5c1def8
modules/rtp_rtcp/include folder cleared of lint warnings
by danilchap
· 9 years ago
796cfaf
Add VideoCodec::PreferDecodeLate
by perkj
· 9 years ago
4d68208
Reduce the runtime of some ACM tests in modules_tests
by Henrik Lundin
· 9 years ago
c490e01
Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to
by nisse
· 9 years ago
b8b6fbb
lint build/include errors fixed in rtp_rtcp module
by danilchap
· 9 years ago
90b9fc9
Roll chromium_revision a02d286..2e451bf (364268:364330)
by kjellander
· 9 years ago
866df66
Typo fix: Enable a bunch of tests that were accidentally disabled
by kwiberg
· 9 years ago
5811a39
Replace EventWrapper in video/, test/ and call/.
by Peter Boström
· 9 years ago
Next »