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webrtc_m130/media/base
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Tim Haloun 6ca98363ee Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created. 
Bug: webrtc:5008
Change-Id: Iaf4788ef2170fa67a8cdee6e9ea6b8c158f286cb
Reviewed-on: https://webrtc-review.googlesource.com/c/92940
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tim Haloun <thaloun@google.com>
Cr-Commit-Position: refs/heads/master@{#24963}
2018-10-04 00:03:10 +00:00
..
adaptedvideotracksource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
adaptedvideotracksource.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
audiosource.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
codec_unittest.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
device.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakeframesource.cc
Add sanity checks to UpdateDelayStatistics and patch unit tests.
2018-09-24 23:13:02 +00:00
fakeframesource.h
Add sanity checks to UpdateDelayStatistics and patch unit tests.
2018-09-24 23:13:02 +00:00
fakemediaengine.h
Add support for send_encodings parameters in addTransceiver
2018-10-01 22:56:30 +00:00
fakenetworkinterface.h
Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created.
2018-10-04 00:03:10 +00:00
fakertp.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertp.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakevideocapturer.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakevideocapturer.h
Introduce TaskQueueForTest.
2018-02-14 15:32:49 +00:00
fakevideorenderer.cc
Enable clang::find_bad_constructs for sdk/ (part 1).
2018-07-26 12:16:31 +00:00
fakevideorenderer.h
Enable clang::find_bad_constructs for sdk/ (part 1).
2018-07-26 12:16:31 +00:00
h264_profile_level_id.cc
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
h264_profile_level_id.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
mediachannel.cc
Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created.
2018-10-04 00:03:10 +00:00
mediachannel.h
Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created.
2018-10-04 00:03:10 +00:00
mediaconfig.h
Remove wire-up for cancelled experement on VAAPI VP8 encoding
2018-06-15 10:04:07 +00:00
mediaconstants.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediaconstants.h
Add RRTR parameter to media engine and pass it to video receive stream
2018-04-06 11:15:12 +00:00
mediaengine.cc
Add support for RtpEncodingParameters num_temporal_layers.
2018-10-03 07:22:51 +00:00
mediaengine.h
Add support for send_encodings parameters in addTransceiver
2018-10-01 22:56:30 +00:00
rtpdataengine_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
turnutils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
turnutils.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
turnutils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videoadapter_unittest.cc
Add option to only request a frame interval change via OnOutputFormatRequest.
2018-09-07 10:36:27 +00:00
videoadapter.cc
Add option to only request a frame interval change via OnOutputFormatRequest.
2018-09-07 10:36:27 +00:00
videoadapter.h
Add option to only request a frame interval change via OnOutputFormatRequest.
2018-09-07 10:36:27 +00:00
videobroadcaster_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
videocapturer_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocapturer.cc
Enable video adaptation for all screenshare content
2018-08-24 12:02:03 +00:00
videocapturer.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
videocapturerfactory.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videocommon_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videocommon.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocommon.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videosourcebase.cc
Enable and fix chromium clang warnings in sdk/android targets.
2018-04-05 11:22:03 +00:00
videosourcebase.h
Move VideoStreamEncoderInterface to api/.
2018-05-21 19:50:37 +00:00
vp9_profile.cc
Reland "Remove rtc::Optional alias and api:optional target"
2018-07-11 19:02:51 +00:00
vp9_profile.h
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00