admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
brandtr ece4aba64e Generalize FEC unit tests and rename GenerateFec. 
- Rename GenerateFec -> EncodeFec in ForwardErrorCorrection. This naming
  is more consistent with DecodeFec.
- Add appropriate using directives, to reduce clutter in tests.
- Move ConstructMediaPackets to fec_test_helper.{h,cc}. This will help
  future tests of ULPFEC/FlexFEC header formatters.
- Generalize tests in rtp_fec_unittest.cc to typed tests. This will help
  testing ForwardErrorCorrection with both ULPFEC and FlexFEC.

This CL should not impact functionality or performance.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2267393002
Cr-Commit-Position: refs/heads/master@{#14314}
2016-09-21 06:16:36 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
Removed api/rtcstats[report].h pseudonyms of api/stats/rtcstats[report].h
2016-09-20 14:10:19 +00:00
audio
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
base
Revert of Allow the DTLS fingerprint verification to occur after the handshake. (patchset #11 id:200001 of https://codereview.webrtc.org/2163683003/ )
2016-09-20 00:21:00 +00:00
build
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
call
Stop increasing loss-based BWE if no feedback is received.
2016-09-20 12:14:52 +00:00
common_audio
Compensate for the IntelligibilityEnhancer processing delay in high bands
2016-09-20 21:52:08 +00:00
common_video
Reland of Update test code to use I420Buffer when writing pixel data. (patchset #1 id:1 of https://codereview.webrtc.org/2343083002/ )
2016-09-19 07:34:52 +00:00
examples
Merge AppRTCDemoJUnitTest into android_junit_tests target
2016-09-19 15:53:30 +00:00
libjingle
GN: Change group deps to public_deps.
2016-09-15 07:53:34 +00:00
media
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
modules
Generalize FEC unit tests and rename GenerateFec.
2016-09-21 06:16:36 +00:00
p2p
Remove dead code branches from P2PtransportChannel unittest.
2016-09-20 07:23:33 +00:00
pc
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
sdk
GN: Enable rtc_common_config for more targets.
2016-09-18 15:12:36 +00:00
stats
RTCTestStats moved into webrtc/stats/test (rtc_stats_test_utils)
2016-09-19 13:06:05 +00:00
system_wrappers
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
test
Generalize FEC unit tests and rename GenerateFec.
2016-09-21 06:16:36 +00:00
tools
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
video
Do not update stream synchronization if no new video packet has been received since last update (e.g. video muted).
2016-09-17 08:00:04 +00:00
voice_engine
Add voe_cmd_test to voice_engine/BUILD.gn (and remove it from voice_engine.gyp, together with the channel_transport gyp target)
2016-09-15 20:12:10 +00:00
.gitignore
…
BUILD.gn
Reenabled the RtcEventLog unittests
2016-09-19 21:42:10 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Merge AppRTCDemoJUnitTest into android_junit_tests target
2016-09-19 15:53:30 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.