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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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b6e8f2f7a713b1c05786337b4fe2a778f2b8f88a
b6e8f2f
Reland of name OpenH264 frame-type conversion function. (patchset #1 id:1 of https://codereview.webrtc.org/1964913002/ )
by pbos
· 9 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
· 9 years ago
1abf937
Revert of Rename OpenH264 frame-type conversion function. (patchset #2 id:20001 of https://codereview.webrtc.org/1943193003/ )
by Peter Boström
· 9 years ago
79553cb
Using ring buffer for AudioVector in NetEq.
by minyue-webrtc
· 9 years ago
17fa672
Fix AllocationSequence to handle the case when TurnPort stops using shared socket.
by Sergey Ulanov
· 9 years ago
d28db7f
Delete all use of tick_util.h.
by Niels Möller
· 9 years ago
b031a2e
Allow WebRTC to offer receiving capability for 120ms Opus packets.
by minyuel
· 9 years ago
f3995f7
NetEq: Implement Expand::Muted
by henrik.lundin
· 9 years ago
60f6ce2
NetEq: Update stats earlier in the GetAudioInternal call
by henrik.lundin
· 9 years ago
47b17dc
NetEq: Replace timescale_holdoff_ with a Countdown timer
by Henrik Lundin
· 9 years ago
6eaa3a4
_boundingSetToSend moved out of tmmbr_help_ into tmmbn_to_send_
by danilchap
· 9 years ago
db3eea0
Fix codec name logging in ivf_file_writer.cc
by hta
· 9 years ago
5488a21
Roll chromium_revision 58963e5878..5f1d704d67 (391406:392277)
by buildbot
· 9 years ago
aa551e6
Add test annotation to PeerConnectionClientTest.testLoopbackVp9DecodeToTexture test.
by kjellander@webrtc.org
· 9 years ago
2aa8426
Android: Handle SurfaceTextureHelper ctor failure for decoder and capturer
by magjed
· 9 years ago
d040480
rtc::Optional<T>: Don't secretly contain a default-constructed T when empty
by kwiberg
· 9 years ago
e30c272
Revert "Reland of Remove SendPacer from ViEEncoder
by perkj
· 9 years ago
e687f78
Moved the functionality in aec_core_internal.h into other files.
by peah
· 9 years ago
ae28408
Jitter delay now depend on protection mode (FEC/NACK).
by philipel
· 9 years ago
a105987
Convert Vp9 Rtp headers to frame references.
by philipel
· 9 years ago
ba6371e
Delete unused video capture support for cropping, non-square pixels, and ARGB screencast scaling.
by nisse
· 9 years ago
85d5108
Add test annotation to PeerConnectionClientTest.testLoopbackVp9 test.
by kjellander@webrtc.org
· 9 years ago
d939d48
Remove Android x86 compilation trybot from CQ.
by kjellander@webrtc.org
· 9 years ago
e69c37b
Separated the functionalities in the OverdriveAndSuppress
by peah
· 9 years ago
23868b6
Broke apart the functionalities in the SubbandCoherence method in the AEC.
by peah
· 9 years ago
6c9b65a
Made the method PartitionDelay independent of the AEC state.
by peah
· 9 years ago
779e97e
Removed the MIPS optimized code for the comfort noise generation in
by peah
· 9 years ago
8d13c4f
Changed the AEC SubbandCoherence function to not use the full AEC state
by peah
· 9 years ago
d251196
Provide isAudioEnabled flag to control audio unit.
by tkchin
· 9 years ago
8f65cdf
Only generate one CNAME per PeerConnection.
by zhihuang
· 9 years ago
630d9ba
Fixed a crash in Objective-C clients when data channel becomes closed.
by skvlad
· 9 years ago
4f543d0
Remove Scanner usage from CPU Monitor.
by Alex Glaznev
· 9 years ago
c7a6569
Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ )
by pbos
· 9 years ago
82d7862
Change default timestamp to 64 bits in all webrtc directories.
by Honghai Zhang
· 9 years ago
e76db89
Fix BoringSSL license path.
by tkchin
· 9 years ago
dd32486
Bitrate prober now keep track of probing cluster id.
by philipel
· 9 years ago
f2eae33
Corrected bug in checking the third number and added extra checks
by dkirovbroadsoft
· 9 years ago
dc7d0d2
Move, almost, all receive side references to RTP to RtpStreamReceiver.
by mflodman
· 9 years ago
b56069e
Enable NACK for audio even if there are no send streams.
by deadbeef
· 9 years ago
31fec40
Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.org/1909333002/.
by solenberg
· 9 years ago
3a33465
Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer.
by zhihuang
· 9 years ago
44c8a37
Removed the file echo_cancellation_internal.h and moved
by peah
· 9 years ago
cf5b37c
Accept all the media profiles required by JSEP.
by zhihuang
· 9 years ago
39a3670
Rename OpenH264 frame-type conversion function.
by pbos
· 9 years ago
3f08dc6
Introduced the new APM data logging functionality into the AEC echo_cancellation.* API layer.
by peah
· 9 years ago
e84cd2e
Cache a ClientHello received before the DTLS handshake has started.
by deadbeef
· 9 years ago
fac23f0
Tune QP threshold for HW codecs.
by Alex Glaznev
· 9 years ago
600246e
Removed SSRC knowledge from ViEEncoder.
by perkj
· 9 years ago
ef00ec1
Update CPU monitor to use moving averages.
by Alex Glaznev
· 9 years ago
28a4456
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
by Per
· 9 years ago
b49ac78
Revert of Use RC_TIMESTAMP_MODE for OpenH264. (patchset #1 id:1 of https://codereview.webrtc.org/1945763002/ )
by pbos
· 9 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
· 9 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
· 9 years ago
274c1dc
Added flag for FEC for video_loopback.
by philipel
· 9 years ago
73987c9
Run "git cl format --full" on a pair of files with ancient formatting
by kwiberg
· 9 years ago
053f917
Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
by ivoc
· 9 years ago
71af75d
Android GlDrawer: Add frame size as argument to draw functions
by magjed
· 9 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
· 9 years ago
825eb58
Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )
by perkj
· 9 years ago
857c5cc
Remove SendPacer from ViEEncoder
by perkj
· 9 years ago
cfc8e3b
Removed all RTP dependencies from ViEChannel and renamed class.
by mflodman
· 9 years ago
fe4b216
Roll chromium_revision 0b4adfd25e..58963e5878 (390907:391406)
by buildbot
· 9 years ago
3815655
Change aggregation window of aecDivergentFilterFraction to 1 second.
by minyue
· 9 years ago
7dd7ab5
Changed the name of the variable overdriveSm and removed the
by peah
· 9 years ago
55dd708
Support RtpEncodingParameters::active in voice engine.
by Taylor Brandstetter
· 9 years ago
1746179
Reducing neteq sync buffer size.
by minyue
· 9 years ago
4adbbcf
Move ADM Create() method to public interface.
by Peter Boström
· 9 years ago
9bfa106
Change the threshold for external VNR.
by jackychen
· 9 years ago
c4deee4
Use RC_TIMESTAMP_MODE for OpenH264.
by Peter Boström
· 9 years ago
c8fe991
Removing SpatialAudio test code
by henrik.lundin
· 9 years ago
87f8c0d
Adding in objc vars for WebRTC GN config.
by Patrik Höglund
· 9 years ago
b1fb72b
NetEq: Move counting of generated CNG samples from DecisionLogic
by henrik.lundin
· 9 years ago
b46083e
This CL introduces a new data logging functionality
by peah
· 9 years ago
696a802
Re-enable Vp9FlexModeRefCount
by philipel
· 9 years ago
35fdb2a
Log WebRTC.Video.AVSyncOffsetInMs.
by pbos
· 9 years ago
5178ee8
NetEq: Use a BuiltinAudioDecoderFactory to create decoders
by kwiberg
· 9 years ago
ddf1653
Android EGL: Synchronize calls to eglCreateContext
by magjed
· 9 years ago
30f118e
This cl deletes the class webrtc::VideoRendererCallback.
by nisse
· 9 years ago
fc88ffe
Fix allocation size in CricketToJavaI420Frame, taking stride into account.
by nisse
· 9 years ago
35151f3
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
by asapersson
· 9 years ago
5a24637
Do not stop a session unless the candidate of a writable connection belongs to the
by Honghai Zhang
· 9 years ago
5dd42fd
Fixing a segfault that can occur when changing the track of an RtpSender.
by deadbeef
· 9 years ago
acf1431
Removing unused resources from building files.
by minyue
· 9 years ago
376b192
Remove VideoCodingModule::VCMPacketizationCallback
by perkj
· 9 years ago
16ac328
Remove VCMRenderBufferSizeCallback.
by Peter Boström
· 9 years ago
1bffc1d
Rename rtc::Time64 --> rtc::TimeMillis.
by nisse
· 9 years ago
bc75d97
Remove PayloadRouter dependency from ViEEncoder.
by perkj
· 9 years ago
5bd3397
Adding 120 ms frame length support in NetEq.
by minyue
· 9 years ago
e4246b6
Roll chromium_revision 8e44a16c27..0b4adfd25e (390884:390907)
by buildbot
· 9 years ago
7d4a6c3
Adds timeout for audio record thread in Java layer
by henrika
· 9 years ago
53ff70f
Reland "Avoiding overflow in cross correlation in NetEq."
by minyue
· 9 years ago
a017b8e
Remove asapersson from webrtc/modules/utility/OWNERS.
by asapersson
· 9 years ago
d1d96b2
VideoCapturerAndroid: Remove deprecated create function with egl context argument
by Magnus Jedvert
· 9 years ago
1ba8d39
Remove webrtc/stream.h and unutilized inheritance.
by pbos
· 9 years ago
60d7e7c
Roll chromium_revision 927a9e3510..8e44a16c27 (390871:390884)
by buildbot
· 9 years ago
7b94c69
Remove DropOldEvents from list of excluded tests. As of last week, the test no longer exists. (https://codereview.webrtc.org/1687703002/)
by terelius
· 9 years ago
c043052
Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ )
by pbos
· 9 years ago
0a1f14e
Roll chromium_revision 9cd13b2cd1..927a9e3510 (390867:390871)
by buildbot
· 9 years ago
bfefb03
Replace scoped_ptr with unique_ptr everywhere
by kwiberg
· 9 years ago
e319059
Roll chromium_revision c6dd4461f5..9cd13b2cd1 (390857:390867)
by buildbot
· 9 years ago
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