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glaznev abdc857967 Update list of supported Android codecs based on field trial dynamically. 
Currently filed trial value which controls H.264 high profile support is
read once only when factory is created. If field trial value is changed for
the next WebRTC call supported codec list need to be updated as well.

BUG=b/34816463

Review-Url: https://codereview.webrtc.org/2685183004
Cr-Commit-Position: refs/heads/master@{#16543}
2017-02-10 22:40:57 +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
pc
Fixing SDP parsing crash due to invalid port numbers.
2017-02-10 20:35:05 +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.