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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2c599d663d45dd760aac8f0b37c1d65d52c19698
/
video
/
rtp_video_stream_receiver.cc
80ba333
Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix
by Karl Wiberg
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
7172ea1
Don't use old RTCP SR reports for remote clock estimation
by Ilya Nikolaevskiy
· 7 years ago
22ec952
Delete in_order argument to RtpReceiver::IncomingRtpPacket
by Niels Möller
· 7 years ago
c62f6c7
RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs
by Karl Wiberg
· 7 years ago
c856dc2
Convert PayloadUnion from a union to a class, step 2
by Karl Wiberg
· 7 years ago
3b3622f
Delete member VideoReceiveStream::Config::Rtp::ulpfec.
by nisse
· 7 years ago
b0573bc
Reorganize config of RTP header extensions for video receive streams.
by Niels Möller
· 7 years ago
73b60b8
Remove the redundant method GetPayloadSpecifics
by Karl Wiberg
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/video/rtp_video_stream_receiver.cc]
f54573b
Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ )
by nisse
· 7 years ago
ca5706d
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
by nisse
· 7 years ago
bdf3072
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
a895836
Remove typedefs.h from webrtc/ root (part 1)
by solenberg
· 7 years ago
8e7eee0
Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
by nisse
· 7 years ago
a646853
Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ )
by nisse
· 7 years ago
5b4b522
Delete Rtx-related methods from RTPPayloadRegistry.
by nisse
· 7 years ago
35713ea
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
by nisse
· 7 years ago
d4fac69
Unwrap picture ids in the RtpFrameReferencerFinder.
by philipel
· 7 years ago
3c39c01
Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
by nisse
· 7 years ago
5c0f6c6
Use RtxReceiveStream.
by nisse
· 7 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
26e3abb
Reverse |rtx_payload_types| map, and rename.
by nisse
· 7 years ago
8b07305
Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker
by eladalon
· 7 years ago
ba050a6
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
f0f7378
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
cf5d485
Add a flags field to video timing extension.
by sprang
· 7 years ago
3bf97cf
Workaround for PacketBuffer bug.
by philipel
· 7 years ago
c0d481a
Protected streams report RTP messages directly to the FlexFec streams
by eladalon
· 7 years ago
822ff2b
Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates
by eladalon
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
d726a3f
Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ )
by brandtr
· 7 years ago
04f4d12
Implement timing frames.
by ilnik
· 7 years ago
2c9f9f2
Only create H264 frames if there are no gaps in the packet sequence number.
by philipel
· 7 years ago
b4ab381
Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver.
by stefan
· 7 years ago
b1f2ff9
Rename class RtpStreamReceiver --> RtpVideoStreamReceiver.
by nisse
· 7 years ago
[Renamed (88%) from webrtc/video/rtp_stream_receiver.cc]
30e8931
Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.
by nisse
· 7 years ago
3184f8e
Dont request keyframes if the stream is inactive or if we are currently receiving a keyframe.
by philipel
· 7 years ago
2c53b13
Request keyframe if the first received frame is not a keyframe.
by philipel
· 7 years ago
d2ef314
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
by nisse
· 7 years ago
0584331
Delete VieRemb class, move functionality to PacketRouter.
by nisse
· 8 years ago
00d802b
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
by ilnik
· 8 years ago
27c46e2
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
by ilnik
· 8 years ago
774f6b4
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
by ilnik
· 8 years ago
29dbb19
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
by ilnik
· 8 years ago
4fa0c4f
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
by ilnik
· 8 years ago
5721866
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
by ilnik
· 8 years ago
64e739a
Add content type information to Encoded Images and add corresponding RTP extension header.
by ilnik
· 8 years ago
fdbfdc9
Let PacketRouter separate send and receive modules.
by nisse
· 8 years ago
0ffdcc5
Delete unneeded includes of deprecated system_wrappers include files.
by nisse
· 8 years ago
54ca919
Allow padding packet in video streams.
by philipel
· 8 years ago
14adba7
Don't allocate any RTPSender object for a receive only RtpRtcp module.
by nisse
· 8 years ago
cd386eb
Delete support for sending RTCP RPSI and SLI messages.
by nisse
· 8 years ago
dea489f
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
by tommi
· 8 years ago
f284b7f
Remove |running_| state from NackModule to avoid running a tight loop.
by tommi
· 8 years ago
a45102f
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
by philipel
· 8 years ago
38cc1d6
Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
by nisse
· 8 years ago
e525d6a
Revert Make the new jitter buffer the default jitter buffer.
by stefan
· 8 years ago
4709e89
Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
by nisse
· 8 years ago
d44ce05
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
by nisse
· 8 years ago
e5bd702
Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ )
by philipel
· 8 years ago
14245cc
Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ )
by nisse
· 8 years ago
6d4dd59
Always call RemoteBitrateEstimator::IncomingPacket from Call.
by nisse
· 8 years ago
bd26ba7
Only update VCMTiming on every received frame instead of every received packet.
by philipel
· 8 years ago
1474212
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
by brandtr
· 8 years ago
27378f3
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ )
by philipel
· 8 years ago
e497495
Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ )
by kjellander
· 8 years ago
fe2bef3
Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
by brandtr
· 8 years ago
09d6ef0
Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ )
by philipel
· 8 years ago
62d02c3
Unit test out of band H264 SPS,PPS within RtpStreamReceiver.
by johan
· 8 years ago
d2b092f
Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
by johan
· 8 years ago
15389c0
Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor.
by nisse
· 8 years ago
914d49d
Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ )
by kjellander
· 8 years ago
f53d737
H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
by johan
· 8 years ago
04926b8
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
by kjellander
· 8 years ago
f20dd00
Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
by philipel
· 8 years ago
c08c191
Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
by philipel
· 8 years ago
0f0763d
Make the new jitter buffer the default jitter buffer.
by philipel
· 8 years ago
022b54e
Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
by philipel
· 8 years ago
07e276c
Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
by johan
· 8 years ago
f7c6d72
Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled.
by brandtr
· 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
f3feeff
Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
by magjed
· 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
b881254
Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
by magjed
· 8 years ago
56124bd
Send audio and video codecs to RTPPayloadRegistry
by magjed
· 8 years ago
fd5a20f
New jitter buffer experiment.
by philipel
· 8 years ago
e6f98c7
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
by brandtr
· 8 years ago
f1bb476
Simplify {,Set}UlpfecConfig interface.
by brandtr
· 8 years ago
d804895
Rename {,Set}GenericFECStatus to {,Set}UlpfecConfig.
by brandtr
· 8 years ago
d55c3f6
Rename FecReceiver to UlpfecReceiver.
by brandtr
· 8 years ago
b5f2c3f
Rename FecConfig to UlpfecConfig in config.h.
by brandtr
· 8 years ago
1d02d3e
Remove RTC_LOGGED_* macro.
by asapersson
· 8 years ago
71eb61c
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
by magjed
· 8 years ago
97667c7
Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
by magjed
· 8 years ago
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