webrtc_m130

History

emircan 1fd66656b3 Modify profiles for H264 encode SW fallback

We have only Constrained Baseline profile available in SW encoder impl
so modify the profile to that in case  of a fallback

BUG=chromium:735959

Review-Url: https://codereview.webrtc.org/2997913003
Cr-Commit-Position: refs/heads/master@{#19436}

2017-08-22 00:30:58 +00:00

api

iSAC floating-point implementation of the Audio{En,De}coderFactoryTemplate APIs

2017-08-21 13:11:18 +00:00

audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

call

Fix size_t to int in RtpDemuxer.

2017-08-21 20:22:54 +00:00

common_audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

common_video

Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )

2017-08-18 09:51:12 +00:00

examples

Remove dep on system_wrappers for apprtc and use only framework.

2017-08-16 14:28:03 +00:00

logging

Add rtc_event_log_unittest_helper.h to relevant BUILD.gn

2017-08-17 15:32:39 +00:00

media

Modify profiles for H264 encode SW fallback

2017-08-22 00:30:58 +00:00

modules

Fix compile error for the win_msvc_rel bot.

2017-08-21 15:53:47 +00:00

ortc

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

p2p

Add TLS TURN tests.

2017-08-17 23:03:04 +00:00

Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.

2017-08-17 21:10:50 +00:00

rtc_base

Fix a delete type mismatch (deleting a sockaddr_in6* as a sockaddr*)

2017-08-18 07:15:19 +00:00

rtc_tools

Delete old CongestionController class

2017-08-21 10:43:33 +00:00

sdk

Android: Fix synchronization problems in VideoFrame release.

2017-08-21 15:02:58 +00:00

stats

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

system_wrappers

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

test

Delete old CongestionController class

2017-08-21 10:43:33 +00:00

video

Revert of Add Jpeg frame writer for test support. (patchset #12 id:220001 of https://codereview.webrtc.org/2990563002/ )

2017-08-20 19:50:56 +00:00

voice_engine

Revert of Stop silently accepting unsupported flags in test binaries (patchset #5 id:150001 of https://codereview.webrtc.org/2968003003/ )

2017-08-18 14:22:36 +00:00

.gitignore

…

BUILD.gn

Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )

2017-08-17 14:59:54 +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

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

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

config.cc

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

config.h

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +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

Protected streams report RTP messages directly to the FlexFec streams

2017-08-02 14:39:07 +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

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

whitespace.txt

…

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.