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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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020eb85075e5478adab0bee00f7c03cdf57a9fcb
020eb85
Revert of CQ: Remove android_dbg trybot. (patchset #1 id:1 of https://codereview.webrtc.org/1977443002/ )
by kjellander
· 8 years ago
dc4eb8c
Refactoring some tests in peerconnectioninterface_unittest.cc.
by Taylor Brandstetter
· 8 years ago
d8b0109
Fix RTX-configuration test with >2 codecs built.
by Peter Boström
· 8 years ago
d215ade
[rtcp] Remb::Parse updated not to use RTCPUtility
by Danil Chapovalov
· 8 years ago
7173ddd
CQ: Remove android_dbg trybot.
by Henrik Kjellander
· 8 years ago
b1fc54d
Corrected the delay agnostic AEC behavior during periods of silent farend signal.
by peah
· 8 years ago
50b5c3b
Remove ViEEncoder::SetNetworkStatus
by perkj
· 8 years ago
b925306
Add magjed@ and perkj@ as webrtc/examples/ owners
by magjed
· 8 years ago
210dd5c
VideoCapturerAndroid: Ignore erroneous startCaptureOnCameraThread calls instead of crashing
by Magnus Jedvert
· 8 years ago
e38e4f6
IWYU: errno.h in base/logging.h
by mostynb
· 8 years ago
060aa57
VideoCapturerAndroid: Force setDisplayOrientation to 0
by Magnus Jedvert
· 8 years ago
7f216b7
Renames TransportController worker_thread to network_thread.
by Danil Chapovalov
· 8 years ago
3fe372d
Fix all -Wnon-virtual-dtor warnings.
by Henrik Kjellander
· 8 years ago
ad6fc5a
Remove remaining quality-analysis (QM).
by Peter Boström
· 8 years ago
919288f
Clamp number of downscales in QualityScaler.
by Peter Boström
· 8 years ago
33b01f2
Adds network thread to rtc::BaseChannel
by Danil Chapovalov
· 8 years ago
3108fc9
Add config continualGatheringPolicy to the IOS RTCConfiguration.
by Honghai Zhang
· 8 years ago
ec81bcd
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
by perkj
· 8 years ago
2f5ae66
Add root owners to webrtc/OWNERS
by kjellander
· 8 years ago
6ab3db2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 8 years ago
7e3968e
Removed MaxEncodedBytes from AudioEncoder.
by ossu
· 8 years ago
65fc62e
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
8a70714
Modernize variable names
by kwiberg
· 8 years ago
cd6ae66
Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
by Fredrik Solenberg
· 8 years ago
faa78dc
Removed old DtlsIdentityRequestObserver::RequestIdentity function signature
by Henrik Boström
· 8 years ago
db7bd3a
FakeDtlsIdentityStore supporting both RSA and ECDSA.
by Henrik Boström
· 8 years ago
b6e8f2f
Reland of name OpenH264 frame-type conversion function. (patchset #1 id:1 of https://codereview.webrtc.org/1964913002/ )
by pbos
· 8 years ago
4996eaa
Revert of Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://cod… (patchset #1 id:1 of https://codereview.webrtc.org/1944273002/ )
by nisse
· 8 years ago
1abf937
Revert of Rename OpenH264 frame-type conversion function. (patchset #2 id:20001 of https://codereview.webrtc.org/1943193003/ )
by Peter Boström
· 8 years ago
79553cb
Using ring buffer for AudioVector in NetEq.
by minyue-webrtc
· 8 years ago
17fa672
Fix AllocationSequence to handle the case when TurnPort stops using shared socket.
by Sergey Ulanov
· 8 years ago
d28db7f
Delete all use of tick_util.h.
by Niels Möller
· 8 years ago
b031a2e
Allow WebRTC to offer receiving capability for 120ms Opus packets.
by minyuel
· 8 years ago
f3995f7
NetEq: Implement Expand::Muted
by henrik.lundin
· 8 years ago
60f6ce2
NetEq: Update stats earlier in the GetAudioInternal call
by henrik.lundin
· 8 years ago
47b17dc
NetEq: Replace timescale_holdoff_ with a Countdown timer
by Henrik Lundin
· 8 years ago
6eaa3a4
_boundingSetToSend moved out of tmmbr_help_ into tmmbn_to_send_
by danilchap
· 8 years ago
db3eea0
Fix codec name logging in ivf_file_writer.cc
by hta
· 8 years ago
5488a21
Roll chromium_revision 58963e5878..5f1d704d67 (391406:392277)
by buildbot
· 8 years ago
aa551e6
Add test annotation to PeerConnectionClientTest.testLoopbackVp9DecodeToTexture test.
by kjellander@webrtc.org
· 8 years ago
2aa8426
Android: Handle SurfaceTextureHelper ctor failure for decoder and capturer
by magjed
· 8 years ago
d040480
rtc::Optional<T>: Don't secretly contain a default-constructed T when empty
by kwiberg
· 8 years ago
e30c272
Revert "Reland of Remove SendPacer from ViEEncoder
by perkj
· 8 years ago
e687f78
Moved the functionality in aec_core_internal.h into other files.
by peah
· 8 years ago
ae28408
Jitter delay now depend on protection mode (FEC/NACK).
by philipel
· 8 years ago
a105987
Convert Vp9 Rtp headers to frame references.
by philipel
· 8 years ago
ba6371e
Delete unused video capture support for cropping, non-square pixels, and ARGB screencast scaling.
by nisse
· 8 years ago
85d5108
Add test annotation to PeerConnectionClientTest.testLoopbackVp9 test.
by kjellander@webrtc.org
· 8 years ago
d939d48
Remove Android x86 compilation trybot from CQ.
by kjellander@webrtc.org
· 8 years ago
e69c37b
Separated the functionalities in the OverdriveAndSuppress
by peah
· 8 years ago
23868b6
Broke apart the functionalities in the SubbandCoherence method in the AEC.
by peah
· 8 years ago
6c9b65a
Made the method PartitionDelay independent of the AEC state.
by peah
· 8 years ago
779e97e
Removed the MIPS optimized code for the comfort noise generation in
by peah
· 8 years ago
8d13c4f
Changed the AEC SubbandCoherence function to not use the full AEC state
by peah
· 8 years ago
d251196
Provide isAudioEnabled flag to control audio unit.
by tkchin
· 8 years ago
8f65cdf
Only generate one CNAME per PeerConnection.
by zhihuang
· 8 years ago
630d9ba
Fixed a crash in Objective-C clients when data channel becomes closed.
by skvlad
· 8 years ago
4f543d0
Remove Scanner usage from CPU Monitor.
by Alex Glaznev
· 8 years ago
c7a6569
Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ )
by pbos
· 8 years ago
82d7862
Change default timestamp to 64 bits in all webrtc directories.
by Honghai Zhang
· 8 years ago
e76db89
Fix BoringSSL license path.
by tkchin
· 8 years ago
dd32486
Bitrate prober now keep track of probing cluster id.
by philipel
· 8 years ago
f2eae33
Corrected bug in checking the third number and added extra checks
by dkirovbroadsoft
· 8 years ago
dc7d0d2
Move, almost, all receive side references to RTP to RtpStreamReceiver.
by mflodman
· 8 years ago
b56069e
Enable NACK for audio even if there are no send streams.
by deadbeef
· 8 years ago
31fec40
Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.org/1909333002/.
by solenberg
· 8 years ago
3a33465
Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer.
by zhihuang
· 8 years ago
44c8a37
Removed the file echo_cancellation_internal.h and moved
by peah
· 8 years ago
cf5b37c
Accept all the media profiles required by JSEP.
by zhihuang
· 8 years ago
39a3670
Rename OpenH264 frame-type conversion function.
by pbos
· 8 years ago
3f08dc6
Introduced the new APM data logging functionality into the AEC echo_cancellation.* API layer.
by peah
· 8 years ago
e84cd2e
Cache a ClientHello received before the DTLS handshake has started.
by deadbeef
· 8 years ago
fac23f0
Tune QP threshold for HW codecs.
by Alex Glaznev
· 8 years ago
600246e
Removed SSRC knowledge from ViEEncoder.
by perkj
· 8 years ago
ef00ec1
Update CPU monitor to use moving averages.
by Alex Glaznev
· 8 years ago
28a4456
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
by Per
· 8 years ago
b49ac78
Revert of Use RC_TIMESTAMP_MODE for OpenH264. (patchset #1 id:1 of https://codereview.webrtc.org/1945763002/ )
by pbos
· 8 years ago
1aa435c
Reland of Android GlDrawer: Add frame size as argument to draw functions (patchset #1 id:1 of https://codereview.webrtc.org/1950953002/ )
by ivoc
· 8 years ago
1726831
Revert of Android GlDrawer: Add frame size as argument to draw functions (patchset #2 id:20001 of https://codereview.webrtc.org/1948473002/ )
by ivoc
· 8 years ago
274c1dc
Added flag for FEC for video_loopback.
by philipel
· 8 years ago
73987c9
Run "git cl format --full" on a pair of files with ancient formatting
by kwiberg
· 8 years ago
053f917
Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
by ivoc
· 8 years ago
71af75d
Android GlDrawer: Add frame size as argument to draw functions
by magjed
· 8 years ago
c6c00b3
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
by phoglund
· 8 years ago
825eb58
Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )
by perkj
· 8 years ago
857c5cc
Remove SendPacer from ViEEncoder
by perkj
· 8 years ago
cfc8e3b
Removed all RTP dependencies from ViEChannel and renamed class.
by mflodman
· 8 years ago
fe4b216
Roll chromium_revision 0b4adfd25e..58963e5878 (390907:391406)
by buildbot
· 8 years ago
3815655
Change aggregation window of aecDivergentFilterFraction to 1 second.
by minyue
· 8 years ago
7dd7ab5
Changed the name of the variable overdriveSm and removed the
by peah
· 8 years ago
55dd708
Support RtpEncodingParameters::active in voice engine.
by Taylor Brandstetter
· 8 years ago
1746179
Reducing neteq sync buffer size.
by minyue
· 8 years ago
4adbbcf
Move ADM Create() method to public interface.
by Peter Boström
· 8 years ago
9bfa106
Change the threshold for external VNR.
by jackychen
· 8 years ago
c4deee4
Use RC_TIMESTAMP_MODE for OpenH264.
by Peter Boström
· 8 years ago
c8fe991
Removing SpatialAudio test code
by henrik.lundin
· 8 years ago
87f8c0d
Adding in objc vars for WebRTC GN config.
by Patrik Höglund
· 8 years ago
b1fb72b
NetEq: Move counting of generated CNG samples from DecisionLogic
by henrik.lundin
· 8 years ago
b46083e
This CL introduces a new data logging functionality
by peah
· 8 years ago
696a802
Re-enable Vp9FlexModeRefCount
by philipel
· 8 years ago
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