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webrtc
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acbc4e85a16871c84c13e4abf75ae217ea84fdbc
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video
78f8386
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
by nisse
· 7 years ago
642a074
Update thread annotiation macros to use RTC_ prefix
by danilchap
· 7 years ago
eac9d2a
Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )
by kjellander
· 7 years ago
3b7353e
Remove typedefs.h from webrtc/ root (part 1)
by solenberg
· 7 years ago
5e02d38
Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
by nisse
· 7 years ago
a9e87e4
Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ )
by nisse
· 7 years ago
4e3045f
Delete Rtx-related methods from RTPPayloadRegistry.
by nisse
· 7 years ago
6ae8262
Add reporting of googContentType via GetStats on send side
by ilnik
· 7 years ago
3fa5b7b
Fix FrameConfigs used for VP8 with four temporal layers.
by sprang
· 7 years ago
2e14b15
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
by nisse
· 7 years ago
5b18967
Move optional.h to webrtc/api/
by kwiberg
· 7 years ago
1a9dee5
Tightening visibility and removing a public_dep.
by mbonadei
· 7 years ago
1f11d1a
Implement googContentType GetStats metric reported on receive side.
by ilnik
· 7 years ago
19cf6be
Unwrap picture ids in the RtpFrameReferencerFinder.
by philipel
· 7 years ago
dfa8f5e
Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
by nisse
· 7 years ago
5eb0c3b
Change reporting of timing frames conditions in GetStats on receive side
by ilnik
· 7 years ago
daf49be
Use RtxReceiveStream.
by nisse
· 7 years ago
36189cd
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
9c6e107
Fix alr tests config
by ilnik
· 7 years ago
054582b
Replace remaining gflags usages with rtc_base/flags
by oprypin
· 7 years ago
00ed864
Recently we moved webrtc/base to webrtc/rtc_base, so these
by mbonadei
· 7 years ago
42e4711
Piggybacking simulcast id and ALR experiment id into video content type extension.
by ilnik
· 7 years ago
47db89a
Add SentToInputFpsRatioPercent UMA metric on send side
by ilnik
· 7 years ago
878f987
Add a new frame generator that cycles through randomly generated slides.
by erikvarga
· 7 years ago
0a6becd
Replace gflags usages with rtc_base/flags in all targets based on test_main
by oprypin
· 7 years ago
5bf39fc
Removing dependencies on stub headers within WebRTC.
by mbonadei
· 7 years ago
a2c4b05
Update jpeg writer to compile on iOS and document it better
by ilnik
· 7 years ago
c205a94
Now that https://codereview.webrtc.org/3003643002 is landed we can
by mbonadei
· 7 years ago
beb74d6
Update video_replay tool to be able to dump .jpg files.
by philipel
· 7 years ago
3ef80a2
Reverse |rtx_payload_types| map, and rename.
by nisse
· 7 years ago
7f62201
Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker
by eladalon
· 7 years ago
775b7c5
Fix places that trigger no-unused-lambda-capture - change to using static-constexpr.
by eladalon
· 7 years ago
ed56bdc
Make CodecType conversion functions non-optional.
by kthelgason
· 7 years ago
3ba7282
Delete unneeded includes of atomic32.h.
by nisse
· 7 years ago
acbfa68
Add experiment to disable ulpfec.
by stefan
· 7 years ago
6a5ac8f
Report max interframe delay over window insdead of interframe delay sum
by ilnik
· 7 years ago
7d8db23
Fix places that trigger no-unused-lambda-capture
by eladalon
· 7 years ago
3a071f6
Let Call::OnRecoveredPacket parse RTP header extensions.
by brandtr
· 7 years ago
de7e264
Remove WebRTC-videocontenttypeextension field trial completely
by ilnik
· 7 years ago
7f56e63
Ignore inter-frame delay stats samples when stream is inactive
by sprang
· 7 years ago
53431d2
Move PacedSender ownership to RtpTransportControllerSend.
by Stefan Holmer
· 7 years ago
80d7b62
Reland of Add Jpeg frame writer for test support.
by ilnik
· 7 years ago
f0c86c0
Move video send/receive stream headers to webrtc/call.
by aleloi
· 7 years ago
0f007ea
Use SingleThreadedTaskQueue in DirectTransport
by eladalon
· 7 years ago
aff1234
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
by philipel
· 7 years ago
fd5a68e
Revert of Add Jpeg frame writer for test support. (patchset #12 id:220001 of https://codereview.webrtc.org/2990563002/ )
by charujain
· 7 years ago
1dae68b
Add Jpeg frame writer for test support.
by ilnik
· 7 years ago
21950c6
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2995153002/ )
by philipel
· 7 years ago
cee6642
Change ThreadChecker to SequencedTaskChecker in VideoReceiveStream
by eladalon
· 7 years ago
0df2c66
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
by sprang
· 7 years ago
90a1756
Reland of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:1 of https://codereview.webrtc.org/2995173002/ )
by emircan
· 7 years ago
7f88afe
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #2 id:20001 of https://codereview.webrtc.org/2925253002/ )
by emircan
· 7 years ago
cc76c33
Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ )
by tkchin
· 7 years ago
7647da7
Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.
by asapersson
· 7 years ago
de39669
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
by asapersson
· 7 years ago
e8a5ec5
Reland of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2999893002/ )
by stefan
· 7 years ago
9db127b
Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ )
by stefan
· 7 years ago
0a27bbc
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
by Niels Möller
· 7 years ago
786989c
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
by emircan
· 7 years ago
de6c74c
Replace absolute path with relative path for GN files.
by Jianjun Zhu
· 7 years ago
a58c48e
Add a flags field to video timing extension.
by sprang
· 7 years ago
62d8f35
Fix incorrect InterframeDelayMaxInMs histogram metrics
by sprang
· 7 years ago
a562a59
Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
by stefan
· 7 years ago
e7df174
Revert of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2998753002/ )
by stefan
· 7 years ago
4a604c8
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
by philipel
· 7 years ago
e1dcdc7
Make the acceptable queue in the cwnd experiment configurable.
by stefan
· 7 years ago
1ac8f7d
Fix (1) EndToEndTest.InitialProbing and (2) EndToEndTest.TriggerMidCallProbing
by eladalon
· 7 years ago
d8a48fb
Workaround for PacketBuffer bug.
by philipel
· 7 years ago
f0920fe
Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ )
by deadbeef
· 7 years ago
b0be841
Add functionality which limits the number of bytes on the network.
by stefan
· 7 years ago
6ab33cf
Wire up RTP keep-alive in ortc api.
by sprang
· 7 years ago
504c822
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
by srte
· 7 years ago
e9507d1
Fix the video buffer size should take rtt into consideration
by gustavogb
· 7 years ago
6e1beec
Request keyframes more frequently on stream start/decoding error.
by philipel
· 7 years ago
cb712cb
Add an experiment for stricter pacing and ALR probing.
by stefan
· 7 years ago
63b0b01
Renamed fields in common_types.h/RtcpStatistics.
by srte
· 7 years ago
783ce68
Rename ViEEncoder to VideoStreamEncoder
by mflodman
· 7 years ago
92bdb64
Tracking mock_process_thread with a GN target
by mbonadei
· 7 years ago
2c1cbc1
Protected streams report RTP messages directly to the FlexFec streams
by eladalon
· 7 years ago
a207a3a
Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates
by eladalon
· 7 years ago
2f3edcc
Remove NullReceiveStatistics
by danilchap
· 7 years ago
5fb3e75
Only one implementation of MockRtpPacketSink once
by eladalon
· 7 years ago
708d0cd
Simplify FakeReceiveStatistics in video send stream tests
by danilchap
· 7 years ago
78133f0
Report minimum PSNR in VideoQualityTest and save corresponding frame to file
by ilnik
· 7 years ago
0667476
Remove traces from {send,receive}_statistics_proxy.cc
by ehmaldonado
· 7 years ago
5410662
Remove setting configuration parameter to itself.
by danilchap
· 7 years ago
2bc62f3
Remove remains of webrtc/base
by ehmaldonado
· 7 years ago
34cf86e
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
by ilnik
· 8 years ago
00a0701
Let alr detector use a budged to detect underuse.
by tschumim
· 8 years ago
59d5575
Use relative paths in GN files.
by jianjun.zhu
· 8 years ago
bc05a6a
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
by sprang
· 8 years ago
2b6f358
Fix video_replay tool to respect RTX stream and fix default parameters.
by ilnik
· 8 years ago
440909e
Fix flaky test VideoSendStreamTest.SendsKeepAlive
by sprang
· 8 years ago
4f870fc
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
by ehmaldonado
· 8 years ago
c6c814d
Remove remains of webrtc/base
by ehmaldonado
· 8 years ago
a5b6c52
Remove webrtc::VideoEncoderFactory
by magjed
· 8 years ago
4b941f0
Report interframe delay sum in old GetStats
by ilnik
· 8 years ago
76de83e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
e662d12
Implement RTP keepalive in native stack.
by sprang
· 8 years ago
37342b9
Report timing frames info in GetStats.
by ilnik
· 8 years ago
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