platform-external-webrtc

Files

asapersson 6b463faccb Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.

Error resilience is currently always enabled for VP9 which reduces quality.

Reland of https://codereview.webrtc.org/2532053002

BUG=webrtc:6783

Review-Url: https://codereview.webrtc.org/2925253002
Cr-Commit-Position: refs/heads/master@{#19385}

2017-08-17 14:28:10 +00:00

api

L16 implementation of the Audio{En,De}coderFactoryTemplate APIs

2017-08-16 13:12:57 +00:00

audio

Replace absolute path with relative path for GN files.

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

call

Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ )

2017-08-16 15:16:25 +00:00

common_audio

Replace absolute path with relative path for GN files.

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

common_video

Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )

2017-08-15 19:31:23 +00:00

examples

Remove dep on system_wrappers for apprtc and use only framework.

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

logging

Replace CHECK(x && y) with two separate CHECK() calls

2017-08-10 00:22:01 +00:00

media

Delete unneeded Start and Stop methods on FlexfecReceiveStream.

2017-08-16 09:41:27 +00:00

modules

Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.

2017-08-17 14:28:10 +00:00

ortc

Wire up RTP keep-alive in ortc api.

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

p2p

Replace CHECK(x && y) with two separate CHECK() calls

2017-08-10 00:22:01 +00:00

Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ )

2017-08-17 13:50:32 +00:00

rtc_base

Revert of Make it possible for tests to set up trace event handlers. (patchset #4 id:60001 of https://codereview.webrtc.org/3002663002/ )

2017-08-17 13:07:02 +00:00

rtc_tools

Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.

2017-08-16 09:48:33 +00:00

sdk

Fix a reference leak when converting I420 VideoFrame to I420Frame.

2017-08-17 10:56:58 +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

Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names

2017-08-17 12:31:02 +00:00

video

Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.

2017-08-17 14:28:10 +00:00

voice_engine

Add thread annotations to AudioLevel

2017-08-10 12:01:21 +00:00

.gitignore

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BUILD.gn

Revert of Make it possible for tests to set up trace event handlers. (patchset #4 id:60001 of https://codereview.webrtc.org/3002663002/ )

2017-08-17 13:07:02 +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

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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.