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ce5570e54ebd03cf51f08ccd25dd02b4eccd2900
platform-external-webrtc/webrtc
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henrik.lundin ce5570e54e Move neteq_rtpplay.cc inside webrtc::test namespace 
This simplifies the code.

BUG=webrtc:2692

Review-Url: https://codereview.webrtc.org/2006723002
Cr-Commit-Position: refs/heads/master@{#12873}
2016-05-24 13:15:03 +00:00
..
api
Move H264BitstreamParser to video_coding.
2016-05-24 10:16:39 +00:00
audio
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
2016-05-11 13:01:19 +00:00
base
Revert of Change initial DTLS retransmission timer from 1 second to 50ms. (patchset #2 id:20001 of https://codereview.webrtc.org/1981463002/ )
2016-05-24 08:49:46 +00:00
build
Remove xmllite and xmpp dependencies from merge_libs.gyp.
2016-05-24 12:46:25 +00:00
call
Improved error checking for file errors in RtcEventLog.
2016-05-24 08:04:02 +00:00
common_audio
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
2016-05-24 12:49:10 +00:00
common_video
Eliminate use of IsZeroSize method
2016-05-24 06:36:00 +00:00
examples
UI for AppRTC Android demo that doesn't require Design Support Library
2016-05-24 07:18:18 +00:00
libjingle
Read recv timestamps from socket (posix only).
2016-05-23 16:19:37 +00:00
media
Remove xmllite and xmpp dependencies from media.gyp
2016-05-24 09:33:55 +00:00
modules
Move neteq_rtpplay.cc inside webrtc::test namespace
2016-05-24 13:15:03 +00:00
p2p
Fixing the behavior of the candidate filter with pooled candidates.
2016-05-23 23:02:29 +00:00
pc
Do not create a temporary transport channel when using max-bundle
2016-05-18 00:49:58 +00:00
sdk
iOS: Replace find with actual filenames listed.
2016-05-24 05:13:51 +00:00
system_wrappers
Adjust max number of samples that can be stored for a histogram (2000->300).
2016-05-24 13:03:45 +00:00
test
Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
2016-05-20 13:29:50 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Add class which periodically computes statistics.
2016-05-23 13:08:05 +00:00
voice_engine
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
2016-05-24 12:49:10 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
…
BUILD.gn
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
…
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +00:00
transport.h
…
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
…
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
UI for AppRTC Android demo that doesn't require Design Support Library
2016-05-24 07:18:18 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Separate out xml dependencies and tests from rtc_unittests into a separate target.
2016-05-24 12:48:02 +00:00
webrtc.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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.