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nisse 891419f8e8 Treat negative ntp time as unset. 
The video send pipeline uses the magic value 0 for an unset ntp time.
However, the receive pipeline uses the magic value -1 for unset
(unclear where, it seems it behaved differently a few months ago).

This makes cl https://codereview.webrtc.org/2469993003/ fail the
P2PTestConductor.ForwardVideoOnlyStream, because that cl removes code
which always clears the ntp time, and enables propagation of ntp time
from the receive pipeline to the send pipeline.

Treating ntp time <= 0 as unset is a small improvement. Ultimately, a
VideoFrame shouldn't carry an ntp time at all.

BUG=webrtc:5740,webrtc:6977

Review-Url: https://codereview.webrtc.org/2620383005
Cr-Commit-Position: refs/heads/master@{#16035}
2017-01-12 18:02:22 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Replace ASSERT by RTC_DCHECK in all non-test code.
2017-01-12 13:15:36 +00:00
audio
GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.
2017-01-05 14:03:24 +00:00
base
Replace ASSERT by RTC_DCHECK in all non-test code.
2017-01-12 13:15:36 +00:00
build
Remove lint ignores and add WrongCall ignore directly to the test.
2017-01-12 14:19:40 +00:00
call
Clean up ramp-up tests and make sure they all pass.
2017-01-09 12:19:24 +00:00
common_audio
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Remove lint ignores and add WrongCall ignore directly to the test.
2017-01-12 14:19:40 +00:00
logging
Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
2016-12-20 13:03:58 +00:00
media
Replace ASSERT by RTC_DCHECK in all non-test code.
2017-01-12 13:15:36 +00:00
modules
Rename fec_tables_xor.h -> fec_rate_table.h
2017-01-12 14:16:24 +00:00
p2p
Replace ASSERT by RTC_DCHECK in all non-test code.
2017-01-12 13:15:36 +00:00
pc
Replace ASSERT by RTC_DCHECK in all non-test code.
2017-01-12 13:15:36 +00:00
sdk
Fix JNI reference leak in MediaCodecVideoEncoder
2017-01-12 14:50:56 +00:00
stats
RTCMediaStreamTrackStats.ssrcIds removed.
2017-01-11 12:47:23 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Don't detect a new frame if a previous packet is used in a previous frame.
2017-01-11 10:01:56 +00:00
tools
Add method needed to extract frame capture and arrival timestamps from rtc event logs.
2016-12-20 16:51:52 +00:00
video
Treat negative ntp time as unset.
2017-01-12 18:02:22 +00:00
voice_engine
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of "Remove webrtc/libjingle/{xmllite,xmpp}"
2017-01-09 07:02:01 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.