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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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9e1e6c599dc0d911fe1888b8a1788a3e1d73117b
9e1e6c5
Corrected access of null pointer in audioproc_f:
by peah
· 9 years ago
63e6a38
Removes verification of audio parameters on Android
by henrika
· 9 years ago
fded4cc
Roll chromium_revision 5fb8c41aea..a8e17a3031 (438448:438476)
by buildbot
· 9 years ago
0989fbc
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
by nisse
· 9 years ago
7b25166
Fix for left shift of negative value in NetEq.
by ivoc
· 9 years ago
bd6c6fa
Delete method Pathname::url and base/urlencode*
by nisse
· 9 years ago
bb66ec3
Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
by skvlad
· 9 years ago
e0eae3c
This CL adds the basic framework for AEC3 in the audio processing module.
by peah
· 9 years ago
db39742
Delete unused class rtc::RegKey.
by nisse
· 9 years ago
e5dc62a
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails.
by kjellander
· 9 years ago
43c5a97
Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002
by nisse
· 9 years ago
8afbc8c
Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )
by nisse
· 9 years ago
36f74e5
Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
by nisse
· 9 years ago
dd3c811
Roll chromium_revision cfd026f99e..5fb8c41aea (438418:438448)
by buildbot
· 9 years ago
a5073c0
Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS
by Henrik Kjellander
· 9 years ago
80df795
Roll chromium_revision e234d53ddf..cfd026f99e (438369:438418)
by buildbot
· 9 years ago
e26b89c
Roll chromium_revision b571577c64..e234d53ddf (438292:438369)
by buildbot
· 9 years ago
62802a1
Fixing possible crash due to RefCountedChannel assignment operator.
by deadbeef
· 9 years ago
b236257
Fixing integer overflow when parsing bandwidth attribute.
by deadbeef
· 9 years ago
9396a08
Roll chromium_revision 79b1930444..b571577c64 (438242:438292)
by buildbot
· 9 years ago
95aa964
Support external audio mixer in webrtc 2.
by gyzhou
· 9 years ago
7af91dd
Removing "crypto_required" from MediaContentDescription.
by deadbeef
· 9 years ago
00fd520
Roll chromium_revision 047b36f906..79b1930444 (438176:438242)
by buildbot
· 9 years ago
b68cc75
ParseCandidate(): Refactor to fix memcheck false positive.
by hnsl
· 9 years ago
f8b262e
Roll chromium_revision e882052d97..047b36f906 (438143:438176)
by buildbot
· 9 years ago
301fc4a
Update common_audio/smoothing_filter.
by minyue
· 9 years ago
bfcf561
Delete VideoFrame default constructor, and IsZeroSize method.
by nisse
· 9 years ago
46711db
Disable flaky QualityScaler tests for now.
by kthelgason
· 9 years ago
277b250
Refactor "secure bool" into explicit PROTO_TLS.
by hnsl
· 9 years ago
1c4b5bc
Roll chromium_revision 632410c83c..e882052d97 (438112:438143)
by buildbot
· 9 years ago
38b6dbc
Autoroller: Support for rolling individual DEPS entries.
by kjellander
· 9 years ago
ef16e99
Add a gtk3 port of peerconnection_client on Linux
by thomasanderson
· 9 years ago
349092b
Logging basic bad call detection
by palmkvist
· 9 years ago
e381015
Revert of New PeerConnectionInterface::GetStats: No bogus default implementation. (patchset #1 id:1 of https://codereview.webrtc.org/2566143002/ )
by hbos
· 9 years ago
4145989
Roll chromium_revision 2d6dcff9ac..632410c83c (438085:438112)
by buildbot
· 9 years ago
07e276c
Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
by johan
· 9 years ago
4b9ff41
setup_links: Remove mojo and WebKit links.
by kjellander
· 9 years ago
8f23094
New PeerConnectionInterface::GetStats: No bogus default implementation.
by hbos
· 9 years ago
03392d0
Fix for negative shift value in NetEq.
by ivoc
· 9 years ago
921019c
Delete unused class AsyncFile.
by nisse
· 9 years ago
1b72300
Roll chromium_revision e5fe50e808..2d6dcff9ac (437879:438085)
by buildbot
· 9 years ago
6de92f9
Don't allow changing ICE pool size after SetLocalDescription.
by deadbeef
· 9 years ago
25ed435
Implement parsing/serialization of a=bundle-only.
by deadbeef
· 9 years ago
39ce11f
Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ )
by gyzhou
· 9 years ago
f6bcac5
Support external audio mixer in webrtc.
by gyzhou
· 9 years ago
1354901
Making audio network adaptor config proto a JAVA package.
by minyue
· 9 years ago
580df53
Fix header guard in thread_annotations.h.
by noahric
· 9 years ago
e5ba75a
Destroy encoders that fail to InitEncode.
by noahric
· 9 years ago
cb44343
Add SSRC to RtpEncodingParameters for audio.
by deadbeef
· 9 years ago
ccecdd4
Roll chromium_revision 45a928c03f..e5fe50e808 (437857:437879)
by buildbot
· 9 years ago
4f19b2f
Add OWNERS to BWE modules.
by stefan
· 9 years ago
fe793eb
Remove sequenced task checker from FlexfecSender.
by brandtr
· 9 years ago
e54b0c5
Roll chromium_revision 88e7649411..45a928c03f (437837:437857)
by buildbot
· 9 years ago
a9a6d4b
Delete voice_engine_configurations.h
by henrik.lundin
· 9 years ago
ba7e71b
remove googViewLimitedResolution stat
by philipp.hancke
· 9 years ago
d2ce622
Disabling the potentially flaky test
by peah
· 9 years ago
bd44bb0
Fix out of bound reads in ParseIceServerUrl() for various input.
by hnsl
· 9 years ago
b010b8f
Roll chromium_revision 4537fa801e..88e7649411 (437826:437837)
by buildbot
· 9 years ago
65a1e77
Try to deflake VideoSendStream tests with ULPFEC.
by brandtr
· 9 years ago
e448dd5
RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector
by hbos
· 9 years ago
b29b9c8
Replace VideoCaptureDataCallback by VideoSinkInterface.
by nisse
· 9 years ago
99f7bfd
Change MANUAL -> DISABLED for ScreenCapturerIntegrationTest tests
by Henrik Kjellander
· 9 years ago
9e3e0da
Roll chromium_revision d33aa11bc5..4537fa801e (437814:437826)
by buildbot
· 9 years ago
951fe73
Roll chromium_revision 5f112f29f4..d33aa11bc5 (437807:437814)
by buildbot
· 9 years ago
bbfed52
Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves
by ssaroha
· 9 years ago
ae875f2
Roll chromium_revision 05b6f4be7e..5f112f29f4 (437804:437807)
by buildbot
· 9 years ago
2cdec07
Roll chromium_revision 73ac3ff0ec..05b6f4be7e (437797:437804)
by buildbot
· 9 years ago
95bfe2d
Roll chromium_revision f10b6162b6..73ac3ff0ec (437786:437797)
by buildbot
· 9 years ago
ba41428
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
by minyue
· 9 years ago
50bac86
Roll chromium_revision cf064891c2..f10b6162b6 (437785:437786)
by buildbot
· 9 years ago
5c8b757
Roll chromium_revision 211f7ed09d..cf064891c2 (437781:437785)
by buildbot
· 9 years ago
18d592c
Roll chromium_revision 3c957a583f..211f7ed09d (437777:437781)
by buildbot
· 9 years ago
0acf8fa
Roll chromium_revision 8ca234a6e4..3c957a583f (437773:437777)
by buildbot
· 9 years ago
d1a38b5
Implement the "needs-ice-restart" logic for SetConfiguration.
by deadbeef
· 9 years ago
3edec7c
Adding error enum to be used by PeerConnectionInterface methods.
by deadbeef
· 9 years ago
d00ff0b
Roll chromium_revision ba3c4f9491..8ca234a6e4 (437769:437773)
by buildbot
· 9 years ago
78b6dc8
Roll chromium_revision a229e828a6..ba3c4f9491 (437767:437769)
by buildbot
· 9 years ago
fe719ac
Roll chromium_revision 39e086ae3e..a229e828a6 (437763:437767)
by buildbot
· 9 years ago
a1c0f9b
Roll chromium_revision 7b5206fba7..39e086ae3e (437761:437763)
by buildbot
· 9 years ago
f71dbf9
Roll chromium_revision bd86a87396..7b5206fba7 (437742:437761)
by buildbot
· 9 years ago
3910792
Roll chromium_revision b671a5234f..bd86a87396 (437704:437742)
by buildbot
· 9 years ago
a6cc7f1
Roll chromium_revision 71d41b68d1..b671a5234f (437664:437704)
by buildbot
· 9 years ago
8d1649d
MANUAL tests of GDI capturers
by zijiehe
· 9 years ago
c5eefaf
Roll chromium_revision 55c13aa961..71d41b68d1 (437596:437664)
by buildbot
· 9 years ago
82dbd33
Roll chromium_revision 3387c1777b..55c13aa961 (437554:437596)
by buildbot
· 9 years ago
7800cbb
Roll chromium_revision 1289bbe752..3387c1777b (437522:437554)
by buildbot
· 9 years ago
1eac19f
Update version of third_party/gtest-parallel.
by ehmaldonado
· 9 years ago
5493b8a
Remove extra uses of basictypes.h.
by pbos
· 9 years ago
3536463
Only store sequence numbers for media stream in FlexFEC end-to-end test.
by brandtr
· 9 years ago
08ab2ee
Roll chromium_revision b558081912..1289bbe752 (437509:437522)
by buildbot
· 9 years ago
ca87b62
Disable failing perf test on Android.
by kthelgason
· 9 years ago
5025463
Modify JavaToStdString to allow ISO-8859-1 encoded strings.
by minyue
· 9 years ago
5a38836
Implement Theil-Sen's method for fitting a line to noisy data (used in bandwidth estimation).
by terelius
· 9 years ago
02cd4d6
RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.
by hbos
· 9 years ago
d82f512
RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector.
by hbos
· 9 years ago
33ce889
Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
by kthelgason
· 9 years ago
0e738d4
Convert seconds_since_epoch to integer.
by ehmaldonado
· 9 years ago
5ad5de3
During AEC development, it is handy to be able to simulate different
by peah
· 9 years ago
df80fd1
When recreating a call based on an aecdump recording the nearend used
by peah
· 9 years ago
a90799d
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
by asapersson
· 9 years ago
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