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d79f97b542a303ca8b7f9a95d5bda5c605d5c59d
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webrtc
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video
d79f97b
Fixing loopback video test by reconfiguring the encoder to correct size.
by mflodman
· 8 years ago
721d402
Create VideoReceiver with external VCMTiming object.
by philipel
· 8 years ago
df2ceb8
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
by nisse
· 8 years ago
0989fbc
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
by nisse
· 8 years ago
bfcf561
Delete VideoFrame default constructor, and IsZeroSize method.
by nisse
· 8 years ago
349092b
Logging basic bad call detection
by palmkvist
· 8 years ago
07e276c
Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
by johan
· 8 years ago
65a1e77
Try to deflake VideoSendStream tests with ULPFEC.
by brandtr
· 8 years ago
3536463
Only store sequence numbers for media stream in FlexFEC end-to-end test.
by brandtr
· 8 years ago
33ce889
Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
by kthelgason
· 8 years ago
a90799d
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
by asapersson
· 8 years ago
1cd0a0a
Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ )
by kthelgason
· 8 years ago
3e9b133
Bump up scaling limit for MediaCodec.
by kthelgason
· 8 years ago
f7c6d72
Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled.
by brandtr
· 8 years ago
10daf86
Simplify an always true condition.
by nisse
· 8 years ago
1cfbd60
Generalize FlexfecReceiveStream::Config.
by brandtr
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
9aa9688
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
by hta
· 8 years ago
243a0a7
Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
by hta
· 8 years ago
e59647b
This approach passes packetization mode to the encoder as part of
by hta
· 8 years ago
d654a9b
Reduce number of FlexFEC VideoSendStreamTests and lower packet loss.
by brandtr
· 8 years ago
4eb03c7
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
by asapersson
· 8 years ago
1a646ee
Wire up BitrateAllocation to be sent as RTCP TargetBitrate
by sprang
· 8 years ago
5e13d41
Remove limit on how often quality scaling downscales
by kthelgason
· 8 years ago
dd40702
Move VideoDecoder::Create() logic to separate internal video decoder factory
by magjed
· 8 years ago
aa354c9
Rename full_stack.cc to full_stack_tests.cc.
by brandtr
· 8 years ago
93c5d03
Start gathering perf data for VP8 + FlexFEC.
by brandtr
· 8 years ago
13d38fb
Delete all of the video_processing module but the denoiser code.
by nisse
· 8 years ago
076c011
Change unit of logged bitrate stats in bytes/s to bits/s.
by asapersson
· 8 years ago
78b4d56
Relanding "Pass time constant to bwe smoothing filter."
by minyue
· 8 years ago
0245da0
Move ownership of PacketRouter from CongestionController to Call.
by nisse
· 8 years ago
706a45e
Added missing include to fix waterfall compile error.
by hbos
· 8 years ago
0c43f77
Update video histograms that do not have a minimum lifetime limit before being recorded.
by asapersson
· 8 years ago
759e0b7
Fix memory leak in video_coding::PacketBuffer::InsertPacket.
by philipel
· 8 years ago
be74270
Calculate JitterBufferDelayInMs in the new jitter buffer.
by philipel
· 8 years ago
a332877
Remove overhead from video bitrate.
by michaelt
· 8 years ago
876222f
Move usage of QualityScaler to ViEEncoder.
by kthelgason
· 8 years ago
320e45a
Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call.
by asapersson
· 8 years ago
352444f
RTC_[D]CHECK_op: Remove superfluous casts
by kwiberg
· 8 years ago
af476c7
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
by kwiberg
· 8 years ago
80ed35e
Implement periodic bandwidth probing in application-limited region.
by sergeyu
· 8 years ago
266f0a4
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
by philipel
· 8 years ago
6287e82
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
by ossu
· 8 years ago
ceecea4
Pass selected cricket::VideoCodec down to internal H264 encoder
by magjed
· 8 years ago
a8eb756
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 8 years ago
9abbf5a
Pass time constanct to bwe smoothing filter.
by michaelt
· 8 years ago
e69a1a9
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
by magjed
· 8 years ago
d7e6ccb
Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ )
by magjed
· 8 years ago
bdbc4b7
Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
by magjed
· 8 years ago
f3feeff
Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
by magjed
· 8 years ago
5f7226f
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
by asapersson
· 8 years ago
5dfac56
Keep all codec parameters in VideoReceiveStream::Decoder
by magjed
· 8 years ago
a6a699a
Sent bitrate stats are incorrect if FlexFEC is configured:
by asapersson
· 8 years ago
33c81d0
Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ )
by magjed
· 8 years ago
69b627d
Move smoothing filter to common audio and exp_filter to base/analytics.
by minyue
· 8 years ago
b881254
Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
by magjed
· 8 years ago
56124bd
Send audio and video codecs to RTPPayloadRegistry
by magjed
· 8 years ago
3c3aef4
Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ )
by minyue
· 8 years ago
223641f
Reland "Move smoothing filter to common audio".
by minyue
· 8 years ago
0c5a154
Try to deflake VideoSendStream tests with FlexFEC.
by brandtr
· 8 years ago
e2b1501
Start probes only after network is connected.
by Sergey Ulanov
· 8 years ago
f6acc2a
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
by magjed
· 8 years ago
509e4fe
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
by magjed
· 8 years ago
7b9feee
Fix PayloadRouter::OnEncodedImage() to handle errors properly.
by sergeyu
· 8 years ago
eacbaea
Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
by magjed
· 8 years ago
42043b9
Stop using hardcoded payload types for video codecs
by Magnus Jedvert
· 8 years ago
10111bc
Passed AudioMixer to AudioState::Config.
by aleloi
· 8 years ago
b7e7b49
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
by asapersson
· 8 years ago
a62f582
Integrate FlexFEC in video_loopback.
by brandtr
· 8 years ago
dd369c6
Reduce full stack test time to 45 secs and add H264 and FlexFEC.
by brandtr
· 8 years ago
527d347
Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ )
by hta
· 8 years ago
39f9729
Add VideoSendStreamTest for FlexFEC.
by brandtr
· 8 years ago
1293aca
Configure FlexFEC in VideoQualityTest.
by brandtr
· 8 years ago
1e3dfbf
Add FlexFEC end-to-end test.
by brandtr
· 8 years ago
b4af3d6
Remove all references to GYP
by Henrik Kjellander
· 8 years ago
08127a9
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
by Erik Språng
· 8 years ago
906c5dc
Revert of Start probes only after network is connected. (patchset #9 id:240001 of https://codereview.webrtc.org/2458863002/ )
by honghaiz
· 8 years ago
5c99c76
Start probes only after network is connected.
by sergeyu
· 8 years ago
cd188f6
Make SendStatisticsProxy let through FlexFEC packets.
by brandtr
· 8 years ago
43cb716
Add ToString method to AggregatedStats and log stats at the end of a call.
by asapersson
· 8 years ago
841de6a
Add FlexFEC to CallTest.
by brandtr
· 8 years ago
614d5b7
Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/
by magjed
· 8 years ago
e950cad
Wire up FlexfecSender in RTP module and VideoSendStream.
by brandtr
· 8 years ago
fd5a20f
New jitter buffer experiment.
by philipel
· 8 years ago
445fb8f
Use correct define in H264 end-to-end tests.
by brandtr
· 8 years ago
8e75a52
Explicitly use RTX for RED in VideoQualityTest and video_loopback.
by brandtr
· 8 years ago
13ceeea
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
by magjed
· 8 years ago
2814598
Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ )
by hta
· 8 years ago
c63fb3a
Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491933002/ )
by hta
· 8 years ago
e6f98c7
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
by brandtr
· 8 years ago
3bba101
Implement H.264 packetization mode 0.
by hta
· 8 years ago
7fe6db9
Revert of Declare VideoCodec.codec_specific_info private (patchset #5 id:80001 of https://codereview.webrtc.org/2452963002/ )
by hta
· 8 years ago
1369c83
Revert of Issue 2434073003: Extract bitrate allocation ... (patchset #4 id:60001 of https://codereview.webrtc.org/2488833004/ )
by sprang
· 8 years ago
647bf43
Reland of Issue 2434073003: Extract bitrate allocation ...
by sprang
· 8 years ago
7927386
Declare VideoCodec.codec_specific_info private
by hta
· 8 years ago
4bc98d4
Revert of Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ )
by sprang
· 8 years ago
8f46c67
Extract bitrate allocation of spatial/temporal layers out of codec impl.
by sprang
· 8 years ago
79e0588
Set actual transport overhead in rtp_rtcp
by michaelt
· 8 years ago
10cbb46
Fixing config for Audio BWE.
by minyue
· 8 years ago
1743a19
Simplify SetFecParameters signature.
by brandtr
· 8 years ago
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