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gerrit-public.fairphone.software
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platform
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external
/
webrtc
/
8789376cd35e055765a72248a8ad444ea2e9438c
/
talk
3e9ad26
Refactor iOS AppRTC parsing code.
by tkchin@webrtc.org
· 10 years ago
a71bb60
Revert 7750 "Don't reset sequence number for a stream on deactiv..."
by sprang@webrtc.org
· 10 years ago
31f7a0e
Don't reset sequence number for a stream on deactivate/reactivate.
by sprang@webrtc.org
· 10 years ago
2faf7ee
Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection.""
by perkj@webrtc.org
· 10 years ago
58edb83
Add video encoder fps and bitrate statistics to Android AppRTCDemo UI.
by glaznev@webrtc.org
· 10 years ago
0087318
Implement settable min/start/max bitrates in Call.
by pbos@webrtc.org
· 10 years ago
dab5d92
Use mirror image for Android AppRTCDemo local preview.
by glaznev@webrtc.org
· 10 years ago
8562f23
OWNERS: Remove tomasl@ and mallinath@
by kjellander@webrtc.org
· 10 years ago
308e7ff
Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."
by kjellander@webrtc.org
· 10 years ago
2751f2a
This adds an Android apk for running tests on the Java layer of PeerConnection.
by perkj@webrtc.org
· 10 years ago
88d14f4
Remove expensive and unnecessary memory alloc for sending black frames on video
by thorcarpenter@google.com
· 10 years ago
bdcf38c
cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class
by magjed@webrtc.org
· 10 years ago
4591fbd
Use size_t more consistently for packet/payload lengths.
by pkasting@chromium.org
· 10 years ago
edc6e57
Support loopback mode and command line execution
by glaznev@webrtc.org
· 10 years ago
f58b455
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
by magjed@webrtc.org
· 10 years ago
6f6ef72
Add DCHECK to ensure that NetEq's packet buffer is not empty
by henrik.lundin@webrtc.org
· 10 years ago
2176db3
AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land)
by henrika@webrtc.org
· 10 years ago
930e004
Add jmi field for packets discarded due to network error
by guoweis@webrtc.org
· 10 years ago
c72a22c
Add preliminary empty file videoframefactory.cc
by magjed@webrtc.org
· 10 years ago
4ef22d1
Setting Opus FEC as default
by minyue@webrtc.org
· 10 years ago
4ec19e3
Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..."
by tommi@webrtc.org
· 10 years ago
858dbbc
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
by magjed@webrtc.org
· 10 years ago
6a782c2
Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
by henrike@webrtc.org
· 10 years ago
a73d746
Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..."
by magjed@webrtc.org
· 10 years ago
bbd8cad
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
by magjed@webrtc.org
· 10 years ago
ece3890
Report total bitrate for all streams in GetStats.
by pbos@webrtc.org
· 10 years ago
35c1ace
Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..."
by magjed@webrtc.org
· 10 years ago
a1f5b96
Remove unnecessary copying of libjingle resource files.
by kjellander@webrtc.org
· 10 years ago
52da44b
WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
by magjed@webrtc.org
· 10 years ago
312614a
Add jmi field for packets discarded due to network error
by guoweis@webrtc.org
· 10 years ago
6ca6190
Fix a SCTP message reordering issue in datachannel.cc.
by jiayl@webrtc.org
· 10 years ago
8038d42
Follow-up fixes for G722
by henrik.lundin@webrtc.org
· 10 years ago
c492231
Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds.
by henrike@webrtc.org
· 10 years ago
d819803
Wire up DSCP support in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
957e802
Refactor SetDefaultEncoderConfig to work on existing codecs.
by pbos@webrtc.org
· 10 years ago
3c1970f
(Auto)update libjingle 79414100-> 79428003
by buildbot@webrtc.org
· 10 years ago
188d3b2
Enable VP9 video codec support on webrtcvideoengine behind a field trial.
by andresp@webrtc.org
· 10 years ago
f85dbce
Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
by henrik.lundin@webrtc.org
· 10 years ago
d105cc8
Change dummy address to use 0.0.0.0 instead of ::
by perkj@webrtc.org
· 10 years ago
a2ef4fe
Prevent a lot of VideoSendStream reconfigures.
by pbos@webrtc.org
· 10 years ago
82775b1
Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime.
by andresp@webrtc.org
· 10 years ago
5e16066
Reland Volume buttons in AppRTCDemo should affect output audio volume (part I).
by henrika@webrtc.org
· 10 years ago
dced5d7
Revert "Advertise G722 as 8 kHz rather than 16 kHz"
by henrik.lundin@webrtc.org
· 10 years ago
34bda43
(Auto)update libjingle 79326895-> 79329222
by buildbot@webrtc.org
· 10 years ago
e5421e9
Volume buttons in AppRTCDemo should affect output audio volume.
by henrika@webrtc.org
· 10 years ago
fd0efb6
Remove deprecated PeerConnection APIs.
by perkj@webrtc.org
· 10 years ago
19b4741
Removing unused method GetDefaultVideoEncoderConfig.
by andresp@webrtc.org
· 10 years ago
0ef890a
(Auto)update libjingle 79285346-> 79320771
by buildbot@webrtc.org
· 10 years ago
6340acd
AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation.
by mcasas@webrtc.org
· 10 years ago
1dcca40
Advertise G722 as 8 kHz rather than 16 kHz
by henrik.lundin@webrtc.org
· 10 years ago
ee9d61c
This fixes a small memory leak (found using Xcode/Instruments on iOS) in
by tkchin@webrtc.org
· 10 years ago
0bae1fa
Wire up bandwidth stats to the new API and webrtcvideoengine2.
by stefan@webrtc.org
· 10 years ago
a22a628
(Auto)update libjingle 79205306-> 79244016
by buildbot@webrtc.org
· 10 years ago
795d003
(Auto)update libjingle 79200114-> 79205306
by buildbot@webrtc.org
· 10 years ago
8125744
Cleanup RTCVideoRenderer interface.
by tkchin@webrtc.org
· 10 years ago
45ecf4c
(Auto)update libjingle 79169148-> 79192489
by buildbot@webrtc.org
· 10 years ago
8944c9d
AppRTCDemoActivity: use differnet Themes for different API levels
by mcasas@webrtc.org
· 10 years ago
fad9aec
Remove protected files from talk/PRESUBMIT.py.
by pbos@webrtc.org
· 10 years ago
88ef632
Falling back on single-stream on multiple SSRC.
by pbos@webrtc.org
· 10 years ago
b5d045e
ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
by perkj@webrtc.org
· 10 years ago
18de6f9
Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
by tommi@webrtc.org
· 10 years ago
c2dd5ee
Prepare for removal of PeerConnectionObserver::OnError.
by perkj@webrtc.org
· 10 years ago
a663d90
(Auto)update libjingle 79104430-> 79104922
by buildbot@webrtc.org
· 10 years ago
5f38c8d
Android AppRTCDemo improvements:
by glaznev@webrtc.org
· 10 years ago
96a9325
Implement external decoder support in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
2236267
Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
by henrik.lundin@webrtc.org
· 10 years ago
5072e0f
Update Android projects to API level 21.
by kjellander@webrtc.org
· 10 years ago
c2c94a9
Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64
by kjellander@webrtc.org
· 10 years ago
78c222b
Update all .isolate files for the new format.
by kjellander@webrtc.org
· 10 years ago
8a130c1
Update Android projects to API level 20.
by kjellander@webrtc.org
· 10 years ago
b7ed779
Implement conference-mode temporal-layer screencast.
by pbos@webrtc.org
· 10 years ago
3bf3d23
Configure A/V sync in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
2dc6f31
Adapting bitrate according to maxplaybackrate for Opus.
by minyue@webrtc.org
· 10 years ago
14146e4
arm64 iOS build.
by tkchin@webrtc.org
· 10 years ago
50ca986
Improve the logging when a TCP connection is deleted.
by jiayl@webrtc.org
· 10 years ago
8219529
Cleaning up r7562-7567.
by minyue@webrtc.org
· 10 years ago
879fac8
(Auto)update libjingle 78822708-> 78823675
by buildbot@webrtc.org
· 10 years ago
5f73a37
Revert 7563 "before rebase" due to wrong submission
by minyue@webrtc.org
· 10 years ago
c11cc8d
Revert 7564 "to submit" due to wrong submission
by minyue@webrtc.org
· 10 years ago
de386bf
to submit
by minyue@webrtc.org
· 10 years ago
c673bb9
before rebase
by minyue@webrtc.org
· 10 years ago
0b62672
adding default rates
by minyue@webrtc.org
· 10 years ago
776e6f2
Use external VideoDecoders in VideoReceiveStream.
by pbos@webrtc.org
· 10 years ago
1abc146
(Auto)update libjingle 78738075-> 78738103
by buildbot@webrtc.org
· 10 years ago
7998089
ApprtDemo Android: Switch between front and back camera.
by perkj@webrtc.org
· 10 years ago
2623695
Renaming bandwidth to bitrate in webrtcvoiceengine.
by minyue@webrtc.org
· 10 years ago
269fb4b
move xmpp and p2p to webrtc
by henrike@webrtc.org
· 10 years ago
ae694ef
(Auto)update libjingle 78642371-> 78680406
by buildbot@webrtc.org
· 10 years ago
fbd55cb
(Auto)update libjingle 78616359-> 78642371
by buildbot@webrtc.org
· 10 years ago
f15dee6
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
by tommi@webrtc.org
· 10 years ago
243eb8e
Adding setting screen to AppRTCDemo.
by glaznev@webrtc.org
· 10 years ago
068b529
(Auto)update libjingle 78583324-> 78583691
by buildbot@webrtc.org
· 10 years ago
2e7ee4b
Fix the SrtpFilter crash caused by two local offers.
by pthatcher@webrtc.org
· 10 years ago
efc82c2
Implement screencast settings for WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
1732df6
Use flags set by the port allocator.
by braveyao@webrtc.org
· 10 years ago
3f7bcc1
(Auto)update libjingle 78430441-> 78445452
by buildbot@webrtc.org
· 10 years ago
c7ed8db
(Auto)update libjingle 78427027-> 78430441
by buildbot@webrtc.org
· 10 years ago
4709887
Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.
by perkj@webrtc.org
· 10 years ago
c9d6d14
patch from issue 25469004
by pthatcher@webrtc.org
· 10 years ago
8fe75ee
(Auto)update libjingle 78381351-> 78389679
by buildbot@webrtc.org
· 10 years ago
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