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webrtc_m130/modules
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Sebastian Jansson 3d61ab12e6 Adds send time to ReceivedPacket struct. 
Bug: webrtc:10742
Change-Id: I7e83d5ec2e23d1db38d02a0c883466ecdcd387c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141874
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28290}
2019-06-14 15:01:36 +00:00
..
audio_coding
Remove sync buffer length from FilteredCurrentDelayMs.
2019-06-13 09:38:22 +00:00
audio_device
Delete TestAudioDeviceModule methods using rtc::PlatformFile
2019-06-12 15:28:41 +00:00
audio_mixer
Fully qualify googletest symbols.
2019-04-09 17:18:20 +00:00
audio_processing
Change StartAecDump methods to work with FILE* and FileWrapper
2019-06-11 13:43:36 +00:00
bitrate_controller
Removes legacy bitrate controller.
2019-06-11 13:16:05 +00:00
congestion_controller
Removes SimulatedTimeClient
2019-06-13 15:37:10 +00:00
desktop_capture
Reland "Link fewer X11-related libraries"
2019-06-11 07:34:49 +00:00
include
Remove some leftover TODOs for webrtc:10336
2019-06-05 11:21:07 +00:00
pacing
Remove PacedSender::PacketSender interface and use PacketRouter directly
2019-06-12 13:09:04 +00:00
remote_bitrate_estimator
Adds send time to ReceivedPacket struct.
2019-06-14 15:01:36 +00:00
rtp_rtcp
Remove sequence_number from RtpPacketInfo.
2019-06-14 11:21:42 +00:00
third_party
Fix guards for headers in third party
2018-08-07 09:39:06 +00:00
utility
Delete deprecated PlatformThread looping
2019-05-03 08:35:42 +00:00
video_capture
Prepare for splitting the api/video:video_frames build rule.
2019-06-10 11:50:51 +00:00
video_coding
Insert startcodes for H264 single NALU packets.
2019-06-14 13:45:21 +00:00
video_processing
Prepare for splitting the api/video:video_frames build rule.
2019-06-10 11:50:51 +00:00
BUILD.gn
Buffer RTCP feedback messages in RtpVideoStreamReceiver
2019-06-03 12:19:36 +00:00
module_common_types_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
OWNERS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00