Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
f62d107aa7ad6fcf2bacec3c6eb4a16469ed2287
f62d107
Revert opus memcheck suppression
by flim
· 9 years ago
0510331
Drop VideoOptions from VideoSendParameters.
by nisse
· 9 years ago
5a83380
Roll chromium_revision b22628f..334c2c6 (381270:381380)
by kjellander
· 9 years ago
56cf60e
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
by kjellander
· 9 years ago
086f851
Add check_deps rules in DEPS files.
by kjellander@webrtc.org
· 9 years ago
e54467f
Use RTCAudioSessionDelegateAdapter in AudioDeviceIOS.
by tkchin
· 9 years ago
4557d33
Roll chromium_revision e1a2958..b22628f (381201:381270)
by kjellander
· 9 years ago
776593b
Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands
by aluebs
· 9 years ago
f5d4786
SSLCertificate::GetChain: Return scoped_ptr
by kwiberg
· 9 years ago
6021fe2
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
by solenberg
· 9 years ago
6baec03
Port::GetStunMessage: Write to scoped_ptr instead of raw pointer
by kwiberg
· 9 years ago
0540242
Add more conditions for CPU detection in denoiser filter.
by jackychen
· 9 years ago
88dec83
Fixing flaky "TestExpireTime" test.
by Taylor Brandstetter
· 9 years ago
7021b92
introduced rtcp::CommonHeader class
by Danil Chapovalov
· 9 years ago
b58a158
Removed the AudioProcessing dependency in EchoCancellerImpl.
by peah
· 9 years ago
8e3949c
Roll chromium_revision 1a84b14..e1a2958 (381152:381201)
by kjellander
· 9 years ago
80c2cd9
Android: Add more info for createPbufferSurface() exceptions
by magjed
· 9 years ago
253534d
Removed the dependency on AudioProcessingImpl in EchoControlMobileImpl
by peah
· 9 years ago
b8fbb54
Removed the dependency on AudioProcessingImpl in GainControlImpl
by peah
· 9 years ago
f8cdd18
Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs"
by asapersson
· 9 years ago
a1cf366
Handle iOS devices with no rear-facing camera
by hjon
· 9 years ago
e5d5e51
Roll chromium_revision 390847b..1a84b14 (381002:381152)
by kjellander
· 9 years ago
e50872b
Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.
by solenberg
· 9 years ago
c4ec4a2
Add breaks in switch statement to fix AppRTCDemo crash
by hjon
· 9 years ago
a9635b8
Use the right mirroring state when switching cameras in AppRTCDemo.
by hjon
· 9 years ago
8bbbf2c
Rename RTCIceConnectionStateMax to RTCIceConnectionStateCount in Objective-C API.
by hjon
· 9 years ago
5ad6bf1
Roll chromium_revision 6a56b54..390847b (380836:381002)
by kjellander
· 9 years ago
7fb69db
Reland the CL to remove candidates when doing continual gathering
by Honghai Zhang
· 9 years ago
1122dc0
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
fb647a6
Initialize/configure video encoders asychronously.
by Peter Boström
· 9 years ago
4c83c05
Implemented more general version of ForwardDiff/RevereseDiff.
by philipel
· 9 years ago
7a4116a
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
by danilchap
· 9 years ago
31642aa
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
d67717f
Make opus memcheck suppression more generic
by Stefan Holmer
· 9 years ago
3b41170
Remove sparse macros (RTC_HISTOGRAM_*_SPARSE_*) that are no longer used.
by asapersson
· 9 years ago
0dc2316
VideoCapturer: Update interface
by magjed
· 9 years ago
b2a24ec
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
479a04c
Suppress invalid read in Opus.
by Stefan Holmer
· 9 years ago
2875077
Roll chromium_revision db8316d..6a56b54 (380688:380836): MSVS 2015 switch
by kjellander@webrtc.org
· 9 years ago
79858f8
Update iOS AppRTCDemo to use the updated Objective-C API.
by hjon
· 9 years ago
9f987d3
Refactor AVAudioSession intialization code.
by tkchin
· 9 years ago
0ce3bf9
Fix lock behavior on RTCAudioSession.
by tkchin
· 9 years ago
b25345e
Replace scoped_ptr with unique_ptr in webrtc/call/
by kwiberg
· 9 years ago
83f831a
Experiment for the nack module.
by philipel
· 9 years ago
84cc918
Replace scoped_ptr with unique_ptr in talk/
by kwiberg
· 9 years ago
2db1dbb
Remove references to build_with_libjingle and libjingle_java GYP variables.
by kjellander@webrtc.org
· 9 years ago
bad7b09
Update examples GYP to avoid rtc_base_approved warning.
by tkchin
· 9 years ago
d44c077
Revert of Safe numeric library: base/numerics (copied from Chromium) (patchset #11 id:250001 of https://codereview.webrtc.org/1753293002/ )
by Tommi
· 9 years ago
35c5336
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
by Tommi
· 9 years ago
e7ba086
Reconfigure video encoders even when not sending.
by Peter Boström
· 9 years ago
0149e75
Remove the (previosly deprecated) Pass methods
by kwiberg
· 9 years ago
3102294
Replace scoped_ptr with unique_ptr in webrtc/pc/
by kwiberg
· 9 years ago
6f59a4f
Revert of Remove candidates when doing continual gathering (patchset #15 id:560001 of https://codereview.webrtc.org/1648813004/ )
by tommi
· 9 years ago
916c76e
Add new files for VideoSourceBase to roll into Chrome in preparation for implementatin.
by perkj
· 9 years ago
84430da
When doing candidate re-gathering in the same generation, Remove the existing local candidate on the same network
by honghaiz
· 9 years ago
44d7392
Roll chromium_revision 5778d35..db8316d (380596:380688)
by kjellander
· 9 years ago
3ad4bd3
Skinmap improvement.
by jackychen
· 9 years ago
86aabb2
Move BitrateAllocator reference from ViEEncoder to VideoSendStream.
by mflodman
· 9 years ago
a590605
Roll chromium_revision 1ff3458..5778d35 (380437:380596)
by kjellander
· 9 years ago
8842c3e
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
4bf0c71
VCMCodecTimer: Change filter from max to 95th percentile
by magjed
· 9 years ago
43166b8
Add IsAcceptableCipher, use instead of GetDefaultCipher.
by torbjorng
· 9 years ago
737f4b8
Removed the ProcessingComponent class
by peah
· 9 years ago
bfa9711
Removed the dependency in GainControlImpl on the ProcessingComponent class
by peah
· 9 years ago
932fdd9
Roll chromium_revision 549602b..1ff3458 (380317:380437)
by kjellander
· 9 years ago
aac2dea
Changed defaults for CreateAnswer in non-constraint mode
by hta
· 9 years ago
50da1d3
Fixed busy loop in case of partially malformed rtcp packet
by danilchap
· 9 years ago
bb9edbd
Removing dependency of the EchoControlMobileImpl class on ProcessingComponent.
by peah
· 9 years ago
f0dcfe2
Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource.
by perkj
· 9 years ago
a97e3cf
Reland of Android VideoCapturerAndroid: Move stopListening() call to stopCaptureOnCameraThread(): https://codereview.webrtc.org/1763673002/
by magjed
· 9 years ago
1069cac
Tune BWE to be a bit less sensitive to spurious delay events.
by stefan
· 9 years ago
430a9c3
Revert of VideoCapturerAndroid: Use one thread per startCapture()/stopCapture() session (patchset #2 id:60001 of https://codereview.webrtc.org/1763673002/ )
by magjed
· 9 years ago
d72595e
Fix NetEq performance test regression
by henrik.lundin
· 9 years ago
9cbebee
VideoCapturerAndroid: Use one thread per startCapture()/stopCapture() session
by magjed
· 9 years ago
1b3530b
Make rtc::TimestampWrapAroundHandler handle backwards wrapping
by sprang
· 9 years ago
6b4f839
Adds a test for an one-way media PeerConnection.
by hta
· 9 years ago
aac3eb2
Minor ObjC API tweaks.
by tkchin
· 9 years ago
2214647
Roll chromium_revision 2d13c45..549602b (380117:380317)
by kjellander
· 9 years ago
c28f0eb
Roll chromium_revision e8cb0a5..2d13c45 (380046:380117)
by kjellander
· 9 years ago
5de6b75
If MSID is encoded in both ways, make the SSRC-level one take priority.
by Taylor Brandstetter
· 9 years ago
dfc2870
Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ )
by perkj
· 9 years ago
f687d53
Drop the 16kHz sample rate restriction on AECM and zero out higher bands
by Alex Luebs
· 9 years ago
3ecb5c8
Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
by solenberg
· 9 years ago
57ae829
Convert IntelligibilityEnhancer to multi-threaded mode
by Alex Luebs
· 9 years ago
c1e55c7
rtt calculation handles time go backwards
by Danil Chapovalov
· 9 years ago
8886c81
- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
by solenberg
· 9 years ago
16daaa5
Fixed incorrect handling of timestamps in video quality test
by sprang
· 9 years ago
97aacee
Filter out the network in networkmonitor if the linkProperties is null.
by honghaiz
· 9 years ago
07e3d89
Roll chromium_revision 7be4202..e8cb0a5 (379879:380046)
by kjellander
· 9 years ago
0d3eef2
Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it.
by perkj
· 9 years ago
32e0c01
Restore type attributes and remove extraneous nullability annotations for Objective-C Mac build
by Jon Hjelle
· 9 years ago
7e74994
Roll chromium_revision b035ad2..7be4202 (379805:379879)
by kjellander
· 9 years ago
ca8b404
Add tracing to interesting media-related methods.
by Peter Boström
· 9 years ago
13e4339
Filter out network-change event with a null interface name.
by Honghai Zhang
· 9 years ago
1a018dc
Prevent a voice channel from sending data before a source is set.
by Taylor Brandstetter
· 9 years ago
1ae6a45
Android VideoCapturerAndroid: Move stopListening() call to stopCaptureOnCameraThread()
by Magnus Jedvert
· 9 years ago
5ed5ed9
Fix VideoToolbox backgrounding issues.
by tkchin
· 9 years ago
3816bfd
Fix incorrect stride information reported by some HW decoders.
by glaznev
· 9 years ago
295c4c2
Reduce camera freeze timeout to 4 sec.
by glaznev
· 9 years ago
5b830fe
Drop the restriction on same forward and reverse sample rate on the AudioFrame interface of the APM
by Alex Luebs
· 9 years ago
Next »