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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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f1f87203d703445ca52ef77d28b7ef0591f79cab
f1f8720
Split ByteBuffer into writer/reader objects.
by jbauch
· 9 years ago
0d05da7
Rent-A-Codec: Reference count the shared iSAC bandwidth estimation state
by kwiberg
· 9 years ago
4cdbd57
AudioCodingModule: Add methods for injecting external encoder stacks
by kwiberg
· 9 years ago
1d03139
Reland https://codereview.webrtc.org/1802993002/
by solenberg
· 9 years ago
052af4c
autoroller: Use 10 characters for shortened Git hashes.
by kjellander
· 9 years ago
7cc9cc0
New method I420Buffer::Copy.
by nisse
· 9 years ago
cc411c0
Reset the BWE when the network changes.
by Honghai Zhang
· 9 years ago
f8711c0
Adding JNI binding for 'active' field in RTP encodings.
by Taylor Brandstetter
· 9 years ago
0d343fa
Remove unused stuff from AudioFrame:
by solenberg
· 9 years ago
059dadf
Added missing TODOs in the beamformer unit test code.
by peah
· 9 years ago
af2f3dd
Reland: Add IntelligibilityEnhancer support to audioproc_float
by Alejandro Luebs
· 9 years ago
d45b95c
Making new unit test assertions use the standard timeout.
by Taylor Brandstetter
· 9 years ago
dd56fa8
Revert "Add IntelligibilityEnhancer support to audioproc_float"
by Alejandro Luebs
· 9 years ago
98c69a0
Add IntelligibilityEnhancer support to audioproc_float
by Alejandro Luebs
· 9 years ago
f537768
Update QuicTransportChannel to latest version of libquic
by mikescarlett
· 9 years ago
9708e9c
Don't call operator== with scoped_ptr<T> and T*
by kwiberg
· 9 years ago
345807e
Remove calls to rtc::scoped_ptr::accept
by kwiberg
· 9 years ago
57d5a2e
Reland of Added a bitexactness test for the gain controller in the audio processing module.
by peah
· 9 years ago
4079cc3
Remove accidentally readded webrtc/base/sslstreamadapterhelper.cc
by Henrik Kjellander
· 9 years ago
b252856
Remove all uses of the HAVE_CONFIG_H define.
by Henrik Kjellander
· 9 years ago
2f36c23
[rtcp] SenderReport::Parse updated not to use RTCPUtility
by danilchap
· 9 years ago
f816035
Move to x509 v3 as required by the WebRTC draft.
by torbjorng
· 9 years ago
c02c0a7
Remove orphaned files.
by torbjorng
· 9 years ago
ceef046
Added a bitexactness test for the beamformer in the audio processing module
by peah
· 9 years ago
2378212
Android HW decoder: Add support for textures when using EGL 1.0
by magjed
· 9 years ago
9c246c4
Change include in metrics.h (change to use systems_wrappers/include/logging.h, base logging breaks chromium.fyi).
by asapersson
· 9 years ago
ae69b02
Fix typo in FakeAdmTest.TestProcess name.
by Peter Boström
· 9 years ago
58d992e
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
by asapersson
· 9 years ago
6b1968e
Allow passing in strings of length zero to FileWrapper::Write without closing the file.
by terelius
· 9 years ago
d4f6ea7
Re-reland of Added a bitexactness test for the echo canceller in the audio processing module.
by peah
· 9 years ago
85e46a8
Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates
by Per
· 9 years ago
4b0c741
Added a bitexactness test for the intelligibility enhancer in the audio processing module
by peah
· 9 years ago
3a2f7e0
Build dynamic framework with podspec for Objective-C API.
by Jon Hjelle
· 9 years ago
918d015
Revert of Added a bitexactness test for the gain controller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1812433002/ )
by peah
· 9 years ago
3587e33
Roll chromium_revision c656a0e..355fabf (383013:383228)
by kjellander
· 9 years ago
303b3c2
Added the JNI interface to get and set RtpParameters and the maximum bitrate limits.
by skvlad
· 9 years ago
a49dc36
Added a bitexactness test for the gain controller in the audio processing module.
by peah
· 9 years ago
84db6fa
Adding BlockMeanCalculator for AEC.
by minyue
· 9 years ago
7c931ad
Fixed a potential deadlock problem in the AGC
by peah
· 9 years ago
1c2af8e
Avoid clicks when muting/unmuting a voe::Channel.
by solenberg
· 9 years ago
60d5f3f
Don't override curve preferences in BoringSSL.
by David Benjamin
· 9 years ago
027fd8f
Revert of Added a bitexactness test for the echo canceller in the audio processing module. (patchset #2 id:40001 of https://codereview.webrtc.org/1827833006/ )
by guidou
· 9 years ago
7ca142e
ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy.
by perkj
· 9 years ago
9adc91d
Revert of Remove code interfacing legacy openssl. (patchset #3 id:40001 of https://codereview.webrtc.org/1808763002/ )
by Torbjorn Granlund
· 9 years ago
4a3a135
More cleanup of cricket::VideoCapturer
by perkj
· 9 years ago
d61bf80
Removed MediaStreamTrackInterface::set_state
by perkj
· 9 years ago
e29f0e2
Reland of Added a bitexactness test for the echo canceller in the audio processing module.
by peah
· 9 years ago
85fa7d5
Move swap_queue.h to base/
by terelius
· 9 years ago
a4f0788
Delete default_send_ssrc_.
by nisse
· 9 years ago
75a2c23
Roll chromium_revision 004c6f1..c656a0e (382886:383013)
by kjellander
· 9 years ago
a0c44ea
Add 16-bit network id to the candidate signaling.
by honghaiz
· 9 years ago
887a19b
Switch to using EGL 1.0 for rendering and HW codec.
by Alex Glaznev
· 9 years ago
1bd9553
Add visibility flag to GYP.
by tkchin
· 9 years ago
038cea3
Roll chromium_revision 6d73369..004c6f1 (382825:382886)
by kjellander
· 9 years ago
24a62d5
Remove WEBRTC_IOS from RTCPeerConnectionFactory public header.
by tkchin
· 9 years ago
a8415fe
Adding comments about threading around CreatePeerConnectionFactory.
by Taylor Brandstetter
· 9 years ago
f752f85
[rtcp] Pli::Parse updated not to use RTCPUtility
by danilchap
· 9 years ago
2f294b4
Roll chromium_revision 3f0d1a9..6d73369 (382768:382825)
by kjellander
· 9 years ago
7ade7b3
Delete class webrtc::VideoRenderer and its header file.
by nisse
· 9 years ago
1509fa1
Delete cricket::VideoRenderer.
by nisse
· 9 years ago
de31855
Add Mic Toggle button to AppRTCDemo (Android).
by solenberg
· 9 years ago
8f59762
Delete VideoRendererInterface.
by Niels Möller
· 9 years ago
c8f952d
Propagate MediaStreamSource state to video tracks the same way as audio.
by perkj
· 9 years ago
8549ed7
Roll chromium_revision 029c2f4..3f0d1a9 (382621:382768)
by kjellander
· 9 years ago
752f36f
Roll chromium_revision ba603a0..029c2f4 (381748:382621)
by kjellander
· 9 years ago
2df29cb
Remove redefined macros from BitrateAdjuster
by emircan
· 9 years ago
dbe2b87
Adding support for RTCRtpEncodingParameters.active flag.
by deadbeef
· 9 years ago
7a43d25
Make the audio channel communicate network state changes to the call.
by skvlad
· 9 years ago
01bcbd0
Make Android min-resolution rotation-agnostic.
by Peter Boström
· 9 years ago
56036ff
cleanup RTCPSender
by danilchap
· 9 years ago
8d2ade6
Reland of Added a bitexactness test for the echo control mobile in the audio processing module
by peah
· 9 years ago
f3cb49f
Refactor some ObjC API init methods.
by Tze Kwang Chin
· 9 years ago
09c3a1e
Use rtc::scoped_refptr for WebRtcVideoCapturer.
by Peter Boström
· 9 years ago
2943f01
Reland of VCMCodecTimer: Change filter from max to 95th percentile (patchset #1 id:1 of https://codereview.webrtc.org/1808693002/ )
by magjed
· 9 years ago
81cbd92
Revert of Initialize/configure video encoders asychronously. (patchset #4 id:60001 of https://codereview.webrtc.org/1757313002/ )
by Peter Boström
· 9 years ago
60624cd
[iOS] Link with base/maccocoathreadhelper.mm on iOS.
by sdefresne
· 9 years ago
6a85d34
Fixing UpdateLevel function in AEC.
by minyue
· 9 years ago
2cb7341
Moved sequence number specific operations from mod_ops.h
by philipel
· 9 years ago
53cf346
Fix race condition in EventTimerPosix
by sprang
· 9 years ago
e56b99e
Update CPU Monitor to report CPU frequency and battery level.
by Alex Glaznev
· 9 years ago
307a092
Support delayed AudioUnit initialization.
by Tze Kwang Chin
· 9 years ago
bc73fe1
Move build scripts to webrtc/build/ios
by hjon
· 9 years ago
0db3db9
Put config in sync between gyp and gn.
by sdefresne
· 9 years ago
0c4de56
Fix potential crashes in the screen capturer on Mac
by Sergey Ulanov
· 9 years ago
121ac12
Fix some warnings in ObjC code.
by tkchin
· 9 years ago
1d19441
Replace RefCountImpl with rtc::RefCountedObject.
by Peter Boström
· 9 years ago
af510af
Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests.
by nisse
· 9 years ago
c5dabdd
Add support for configuring the number of spatial/temporal layers for VP9 through a field trial.
by asapersson
· 9 years ago
f26f98b
Revert of Added a bitexactness test for the echo canceller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1809613002/ )
by peah
· 9 years ago
b60be20
Revert of Added a bitexactness test for the echo control mobile in the audio processing module (patchset #3 id:60001 of https://codereview.webrtc.org/1805373002/ )
by peah
· 9 years ago
eb83a1a
This is an initial cleanup step, aiming to delete the
by nisse
· 9 years ago
105831e
Added a bitexactness test for the echo control mobile in the audio processing module
by peah
· 9 years ago
7c448e1
Added a bitexactness test for the echo canceller in the audio processing module.
by peah
· 9 years ago
62411a2
Fixing crash that may occur after destroying a VideoSendStream.
by deadbeef
· 9 years ago
bdbceef
Added a bitexactness test for the voice activity detector in the audio processing module.
by peah
· 9 years ago
9e083d2
Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )
by perkj
· 9 years ago
19b7b66
Added a bitexactness test for the level estimator in the audio
by peah
· 9 years ago
caafdba
Fix broken CVO header extension
by perkj
· 9 years ago
eec21bd
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 9 years ago
5585001
Added a bitexactness test for the noise suppressor.
by peah
· 9 years ago
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