webrtc_m130

History

ehmaldonado b0b721a68c Increase the size of the buffer for type.name.id.

BUG=chromium:653087

Review-Url: https://codereview.webrtc.org/2977363002
Cr-Commit-Position: refs/heads/master@{#19072}

2017-07-18 11:27:08 +00:00

api

Remove default implementation of PeerConnectionInterface::SetBitrate.

2017-07-17 22:09:30 +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

Remove potential left shift of negative value in WebRtcSpl_AnalysisQMF

2017-07-18 10:02:28 +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

Patch set 1:

2017-07-18 11:12:29 +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

Increase the size of the buffer for type.name.id.

2017-07-18 11:27:08 +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

Android bindings for ice_regather_interval_range RTCConfiguration option

2017-07-17 20:09:43 +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.