platform-external-webrtc

Files

sakal fcf97c3b75 Fix fullscreen scaling in AppRTCMobile.

The surface view renderer size was set to match parent so it couldn't
adjust based on the frame size. The size is now set to wrap_content
which allows the renderer to adjust. The root element of the call
activity is changed to FrameLayout to allow the renderer to center.

requestLayout is added to SurfaceView setScalingType so onMeasure gets
called again.

BUG=webrtc:7901

Review-Url: https://codereview.webrtc.org/2978173002
Cr-Commit-Position: refs/heads/master@{#19073}

2017-07-18 12:01:08 +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

Fix fullscreen scaling in AppRTCMobile.

2017-07-18 12:01:08 +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

Fix fullscreen scaling in AppRTCMobile.

2017-07-18 12:01:08 +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.