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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
73e1a8bb741e8e3956815f75b4977c99f0bba0db
/
video_engine
73e1a8b
Disable flaky RunsRtpRtcpTestWIthoutErrors.
by pbos@webrtc.org
· 10 years ago
b0079ed
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 10 years ago
0a5fd54
Casting char to int in logs.
by asapersson@webrtc.org
· 10 years ago
b991cd0
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
by wu@webrtc.org
· 10 years ago
a1626fe
Propagate capture ntp timestamp from rtp to renderer.
by wu@webrtc.org
· 10 years ago
6b1114a
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
by fischman@webrtc.org
· 10 years ago
b18bff5
Cleaned up logging in video_coding.
by stefan@webrtc.org
· 10 years ago
4b0cd7f
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
by andresp@webrtc.org
· 10 years ago
ff46b81
Updated WebRTC version to 3.52 TBR=wu@webrtc.org
by elham@webrtc.org
· 10 years ago
44c9b9a
Log Fixit for parts of video_engine folder.
by mflodman@webrtc.org
· 10 years ago
6c57efd
Re-submit: rev5775
by andresp@webrtc.org
· 10 years ago
37f807f
Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.
by solenberg@webrtc.org
· 10 years ago
765ea72
Revert 5775 "Modify bitrate controller to update bitrate based o..."
by andrew@webrtc.org
· 10 years ago
f50914a
Removing VideoCodecDerived and moving methods inside VideoCodec.
by mallinath@webrtc.org
· 10 years ago
0ac0bca
Updated WebRTC version to 3.51
by elham@webrtc.org
· 10 years ago
539bbde
Modify bitrate controller to update bitrate based on process call and not
by andresp@webrtc.org
· 10 years ago
1f49208
Adding API for setting bandwidth estimation configurations.
by stefan@webrtc.org
· 10 years ago
1a19092
Add configuration for ability to use the encode usage measure for triggering overuse/underuse.
by asapersson@webrtc.org
· 10 years ago
50ac4d6
Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API.
by solenberg@webrtc.org
· 10 years ago
85101db
Have changes to REMB trigger RTCP to be sent immediately.
by stefan@webrtc.org
· 10 years ago
5f804f8
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 10 years ago
9b2b8ec
Add AIMD option to BWE API.
by stefan@webrtc.org
· 10 years ago
209791d
Refactor in BitrateController module.
by andresp@webrtc.org
· 10 years ago
40fee00
Adding operator== and != methods for CodecInst and VideoCodec structures.
by mallinath@webrtc.org
· 10 years ago
27bd3be
Add ability to configure cpu overuse options via an API.
by asapersson@webrtc.org
· 10 years ago
f9d5709
Fixes RTX related bugs.
by stefan@webrtc.org
· 10 years ago
bef6e62
Simplify pacer interface.
by pbos@webrtc.org
· 10 years ago
3a70e6e
Fix a deadlock in ViEEncoder::DeliverFrame.
by wuchengli@chromium.org
· 10 years ago
9420a1f
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
f35f098
Avoid crash in ViEEncoder::DeRegisterExternalEncoder().
by fischman@webrtc.org
· 10 years ago
0bf5a2f
Adding a new ramp-up-down-up test
by henrik.lundin@webrtc.org
· 10 years ago
ecee063
Adds APIs for reporting pacer queuing delay.
by jiayl@webrtc.org
· 10 years ago
c0d56c0
Fix to get total number of sent and received rtcp packets.
by asapersson@webrtc.org
· 10 years ago
23c8d6b
Updated WebRTC version to 3.50 TBR= wu@webrtc.org
by elham@webrtc.org
· 10 years ago
072bab2
Modified overuse detection thresholds.
by asapersson@webrtc.org
· 10 years ago
2fa9f7e
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
by asapersson@webrtc.org
· 10 years ago
ae50521
Remove external encryption API for VoE.
by solenberg@webrtc.org
· 10 years ago
15e3511
Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending).
by asapersson@webrtc.org
· 10 years ago
46b22d8
Adding a critical section missing in r5543.
by stefan@webrtc.org
· 10 years ago
8e98655
Increase overuse and normal use thresholds for Mac.
by asapersson@webrtc.org
· 10 years ago
8cb4c8d
Fixes a race when writing to send_padding_.
by stefan@webrtc.org
· 10 years ago
6cfc58d
Set pacing bitrates in SetEncoder.
by pbos@webrtc.org
· 10 years ago
ddbd31e
Remove ViE external encryption API.
by solenberg@webrtc.org
· 10 years ago
a68379b
Add stats of incoming frame delays for debugging bandwidth estimation.
by jiayl@webrtc.org
· 10 years ago
49e9e15
Connect webrtc::Config to WrappingBitrateEstimator
by henrik.lundin@webrtc.org
· 11 years ago
a1e140d
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
by mallinath@webrtc.org
· 11 years ago
c091c50
Revert 5421 "Fix deadlock on register/unregister observer while ..."
by mallinath@webrtc.org
· 11 years ago
aa2c3ae
Fix deadlock on register/unregister observer while there is a an going callback.
by andresp@webrtc.org
· 11 years ago
224933c
Add callbacks for receive channel RTP statistics
by sprang@webrtc.org
· 11 years ago
64339f0
Add configuration and test for extended RTCP reference time reports to new video api.
by asapersson@webrtc.org
· 11 years ago
c1792c5
Roll Chromium 238260 -> 243863
by wjia@webrtc.org
· 11 years ago
b95f445
Updated Webrtc version to 3.49
by elham@webrtc.org
· 11 years ago
9c8f391
Removes usage of ListWrapper from several files.
by henrike@webrtc.org
· 11 years ago
ca72300
Wire up statistics in video send stream of new video engine api
by sprang@webrtc.org
· 11 years ago
7e4053c
Add thread_annotations for clang targets.
by andresp@webrtc.org
· 11 years ago
d3f0617
If the configured start bitrate is higher than the configures max
by mflodman@webrtc.org
· 11 years ago
4a185e9
Race condition in ViECapturer::RegisterObserver
by sprang@webrtc.org
· 11 years ago
e83367b
Update WebRTC to version 3.48
by tnakamura@webrtc.org
· 11 years ago
acc2e43
Add callbacks for receive channel RTCP statistics.
by sprang@webrtc.org
· 11 years ago
cd117d2
Integrate fake_network_pipe into direct_transport.
by stefan@webrtc.org
· 11 years ago
ef1f6c3
Remove media_file from VideoEngine dependencies.
by pbos@webrtc.org
· 11 years ago
39139dc
Revert r5294 to re-roll r5293.
by pbos@webrtc.org
· 11 years ago
0af1d21
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
by turaj@webrtc.org
· 11 years ago
ee867fa
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
by solenberg@webrtc.org
· 11 years ago
0e4512b
Callback for send bitrate estimates - new roll
by sprang@webrtc.org
· 11 years ago
e4d538a
Make sure channels in the same call are in the same channel group.
by mflodman@webrtc.org
· 11 years ago
e6dc4ff
Making RemoteRateControl::min_configured_bit_rate_ configurable
by henrik.lundin@webrtc.org
· 11 years ago
3a4fc4b
Update talk to 58127566 together with
by wu@webrtc.org
· 11 years ago
9b3d2bf
Revert 5274 "Update talk to 58113193 together with https://webrt..."
by wu@webrtc.org
· 11 years ago
cde78d6
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
by wu@webrtc.org
· 11 years ago
a4670a1
Complete rewrite of demo application.
by henrike@webrtc.org
· 11 years ago
0ceb51f
Remove overloaded CpuOveruseMeasure function.
by asapersson@webrtc.org
· 11 years ago
7123a80
Add SwapFrame() to VideoSendStreamInput.
by pbos@webrtc.org
· 11 years ago
66e84b0
Revert 5259 "Callback for send bitrate estimates"
by sprang@webrtc.org
· 11 years ago
894dab9
Roll chromium_revision 232627:238260
by kjellander@webrtc.org
· 11 years ago
f1d22d4
Callback for send bitrate estimates
by sprang@webrtc.org
· 11 years ago
ed8c496
Fraction lost statistics not being reported
by sprang@webrtc.org
· 11 years ago
cf5c552
Add callbacks for send channel rtp statistics
by sprang@webrtc.org
· 11 years ago
8db148e
Add API to query video engine for the send-side delay.
by stefan@webrtc.org
· 11 years ago
adc238a
Fixing the android build
by henrik.lundin@webrtc.org
· 11 years ago
b669e60
Remove default implementations for SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
3bcea52
Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics.
by stefan@webrtc.org
· 11 years ago
8911937
Add callbacks for send channel rtcp statistics
by sprang@webrtc.org
· 11 years ago
5459e0b
Remove the long disabled WEBRTC_SVNREVISION define.
by andrew@webrtc.org
· 11 years ago
382cfdd
Removing DropDeltaAfterKey functionality which is unused.
by andresp@webrtc.org
· 11 years ago
9435a17
Add send frame rate statistics callback
by sprang@webrtc.org
· 11 years ago
f2c136b
Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second.
by asapersson@webrtc.org
· 11 years ago
da3ae7c
Adds support for sending redundant payloads over RTX.
by stefan@webrtc.org
· 11 years ago
0e2571d
Lock frame in ViECapturer::IncomingFrameI420.
by pbos@webrtc.org
· 11 years ago
801822c
Ensure that no packet stays in the pacer queue for longer than 2 seconds.
by stefan@webrtc.org
· 11 years ago
8f9da30
Create default implementation to fix issue in libjingle
by sprang@webrtc.org
· 11 years ago
4a9843f
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
2622be1
Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api.
by asapersson@webrtc.org
· 11 years ago
58b912b
Remove const in vie_rtp_rtcp, where there is conflict with
by sprang@webrtc.org
· 11 years ago
1a5aa03
Updated WebRTC version to 3.47 TBR=wu@webrtc.org
by elham@webrtc.org
· 11 years ago
c86d1c6
Add include stdlib.h to files using abs.
by stefan@webrtc.org
· 11 years ago
44b21e7
Replace VideoFrameI420 with I420VideoFrame.
by pbos@webrtc.org
· 11 years ago
ce4a0b8
Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release.
by dwkang@webrtc.org
· 11 years ago
970c5e5
Renaming ViEEncoderObserver::VideoSuspended
by henrik.lundin@webrtc.org
· 11 years ago
a706baf
Protect reads of ViEEncoder::video_suspended_.
by pbos@webrtc.org
· 11 years ago
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