platform-external-webrtc

Files

deadbeef a4549d6588 Fix SDP parsing crash due to missing track ID in "a=msid".

BUG=chromium:686405

Review-Url: https://codereview.webrtc.org/2676293003
Cr-Commit-Position: refs/heads/master@{#16545}

2017-02-11 01:26:22 +00:00

api

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

audio

Wire up audio packet loss to BWE.

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

base

Fixing SDP parsing crash due to invalid port numbers.

2017-02-10 20:35:05 +00:00

call

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

common_audio

Fix for left shift of potentially negative value.

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

common_video

Remove henrik.lundin from webrtc/common_video/OWNERS

2017-02-10 16:44:31 +00:00

examples

Move Android video quality loopback script.

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

logging

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

media

Rename "PacketTransportInterface" to "PacketTransportInternal".

2017-02-10 19:31:50 +00:00

modules

Fix nr of bytes sent to Opus decoder in DTX mode

2017-02-10 21:50:38 +00:00

p2p

Rename "PacketTransportInterface" to "PacketTransportInternal".

2017-02-10 19:31:50 +00:00

Fix SDP parsing crash due to missing track ID in "a=msid".

2017-02-11 01:26:22 +00:00

sdk

Update list of supported Android codecs based on field trial dynamically.

2017-02-10 22:40:57 +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

Make functions in fileutils.h use "const std::string&".

2017-02-10 17:27:14 +00:00

tools

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

video

Retransmitted packets are now counted in receive time

2017-02-10 17:16:05 +00:00

voice_engine

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

.gitignore

…

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

…

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

…

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

…

PRESUBMIT.py

…

README.chromium

…

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

Check for use_x11 before runnig desktop_capture_modules_tests on linux.

2017-02-10 09:38:23 +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.