platform-external-webrtc

Files

deadbeef faedf7f942 Getting rid of "benign blocking error" log spam.

For UDP sockets, instead of calling "recv" with "MSG_PEEK" to see if the socket
is alive, just check whether or not it's been closed. It seems that, at least
on some versions of Android, with some specific conditions involving loopback
sockets, calling "recv" with a buffer that's too small for a UDP packet causes
an EFAULT error.

BUG=webrtc:6715

Review-Url: https://codereview.webrtc.org/2678353006
Cr-Commit-Position: refs/heads/master@{#16522}

2017-02-09 23:09:22 +00:00

api

Adding more comments to every header file in api/ subdirectory.

2017-02-08 09:38:21 +00:00

audio

Wire up audio packet loss to BWE.

2017-02-07 15:14:08 +00:00

base

Getting rid of "benign blocking error" log spam.

2017-02-09 23:09:22 +00:00

call

Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

2017-02-07 09:18:43 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-02-09 11:05:59 +00:00

common_video

Reland of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #2 id:150001 of https://codereview.webrtc.org/2687073002/ )

2017-02-09 12:58:53 +00:00

examples

Move Android video quality loopback script.

2017-02-09 09:05:54 +00:00

logging

Add option to print information about configured SSRCs from RTC event logs.

2017-02-08 17:37:30 +00:00

media

Added a flag to AudioCodecSpec to indicate adaptive bitrate support.

2017-02-09 13:14:32 +00:00

modules

replace NtpTime->Clock with Clock->NtpTime dependency

2017-02-09 19:15:25 +00:00

p2p

Increase STUN RTOs to work better on poor networks, such as 2G networks.

2017-02-08 22:42:22 +00:00

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

sdk

Roll chromium_revision baaeb3f30e..ba3e8dd8fa (449272:449304)

2017-02-09 19:19:31 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

2017-01-27 14:35:16 +00:00

system_wrappers

replace NtpTime->Clock with Clock->NtpTime dependency

2017-02-09 19:15:25 +00:00

test

Add new target direct_transport and remove fake_network and direct_transport from test_common.

2017-02-08 13:55:51 +00:00

tools

Move Android video quality loopback script.

2017-02-09 09:05:54 +00:00

video

Fixing typo

2017-02-09 16:32:53 +00:00

voice_engine

Remove unused voe_stress_test.cc

2017-02-09 13:03:25 +00:00

.gitignore

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BUILD.gn

Added gn target for rtc_event_log2rtp_dump.

2017-02-08 08:28:09 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

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common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

config.cc

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

config.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

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LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

no_op_function.cc

Moving no_op_function.cc out of webrtc/build

2017-01-24 13:25:12 +00:00

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Only define NO_RETURN if undefined

2017-01-30 08:54:19 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

webrtc.gni

Adding build switch for Opus that supports 120ms ptime.

2017-02-02 01:31:11 +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.