platform-external-webrtc

Files

Zijie He 44967e41c5 Expose IsCurrentSessionSupported() from ScreenCapturerWinDirectx

IsCurrentSessionSupported() is useful to decide whether Windows version should
be used to evaluate the capability of DirectX capturer on the system.

Bug: 741926
Change-Id: Iaaf6011a9e464d7cf5e7dda097007778c73953e0
Reviewed-on: https://chromium-review.googlesource.com/571378
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19060}

2017-07-17 18:58:03 +00:00

api

Reinstate "API for periodically regathering ICE candidates"

2017-07-14 22:26:05 +00:00

audio

Don't run NoBandwidthDropAfterDtx test on andriod because it's flaky.

2017-07-17 15:19:27 +00:00

base

Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )

2017-07-11 23:56:05 +00:00

call

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy .

2017-07-14 19:17:49 +00:00

common_audio

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

common_video

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

examples

Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )

2017-07-11 23:56:05 +00:00

logging

nit: Avoid pointer-to-unique_ptr in RtcEventLogImpl

2017-07-16 23:44:08 +00:00

media

Revert "Prefer external video codecs over internal in SDP"

2017-07-17 15:45:17 +00:00

modules

Expose IsCurrentSessionSupported() from ScreenCapturerWinDirectx

2017-07-17 18:58:03 +00:00

ortc

Disabling test on iOS64 debug bot

2017-07-14 22:19:03 +00:00

p2p

Reinstate "Add additional check when setting RTCConfiguration"

2017-07-14 23:40:53 +00:00

Reinstate "Add additional check when setting RTCConfiguration"

2017-07-14 23:40:53 +00:00

rtc_base

Reinstate "API for periodically regathering ICE candidates"

2017-07-14 22:26:05 +00:00

rtc_tools

Added -show_detector_state which show the detector state in the total bitrate graph.

2017-07-14 13:30:03 +00:00

sdk

Revert of Injectable Obj-C video codecs (patchset #2 id:370001 of https://codereview.webrtc.org/2979983002/ )

2017-07-15 02:49:58 +00:00

stats

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy .

2017-07-14 19:17:49 +00:00

system_wrappers

Fix DetectNumberOfCores for 32-bit processes

2017-07-13 17:06:12 +00:00

test

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy .

2017-07-14 19:17:49 +00:00

video

Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback

2017-07-13 07:47:03 +00:00

voice_engine

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy .

2017-07-14 19:17:49 +00:00

.gitignore

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BUILD.gn

Reimplemeted "Test and fix for huge bwe drop after alr state"

2017-07-17 08:41:41 +00:00

codereview.settings

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common_types.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

common_types.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

DEPS

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

LICENSE

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LICENSE_THIRD_PARTY

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OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Report interframe delay sum in old GetStats

2017-07-07 08:26:24 +00:00

video_send_stream.h

Move RTP keep-alive config from VideoSendStream::Config to Call::Config

2017-07-11 10:44:17 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +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.