platform-external-webrtc

Files

Magnus Jedvert 0706813280 Android AppRTCMobile: Add setting for selecting H264 Baseline or High profile

BUG=webrtc:6337
R=glaznev@webrtc.org

Review-Url: https://codereview.webrtc.org/2658243002 .
Cr-Commit-Position: refs/heads/master@{#16605}

2017-02-14 11:41:35 +00:00

api

Changing some PeerConnection-related comments.

2017-02-13 17:49:58 +00:00

audio

Clean out platform specific things from voice_engine_defines.h.

2017-02-13 12:42:52 +00:00

base

Delete httpclient.c and related files.

2017-02-14 08:23:10 +00:00

call

Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

2017-02-13 13:59:46 +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

Android AppRTCMobile: Add setting for selecting H264 Baseline or High profile

2017-02-14 11:41:35 +00:00

logging

Move AudioDecoder and related stuff to the api/ directory

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

media

Handle InitDecode and reset in fallback decoder.

2017-02-13 22:11:08 +00:00

modules

Added handling of 'agc_compression_gain' flag in audioproc_f.

2017-02-14 10:07:49 +00:00

p2p

Fix the build break by keeping the old Port::AddAddress method since the

2017-02-14 01:33:27 +00:00

Changing some PeerConnection-related comments.

2017-02-13 17:49:58 +00:00

sdk

Compile ios helpers on mac as well.

2017-02-14 11:07:57 +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

Do not encode frames in MultithreadedFakeH264Encoder after Release().

2017-02-11 08:25:18 +00:00

tools

Move AudioDecoder and related stuff to the api/ directory

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

video

Make sure initial framedrop is off where quality scaling is off.

2017-02-14 08:46:51 +00:00

voice_engine

Clean out platform specific things from voice_engine_defines.h.

2017-02-13 12:42:52 +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

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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

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.