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hta 41286496cb Move all codec specific definitions from modules_include 
This CL moves all codec specific definitions into their own
header files in the respective codec directory, and replaces
it with an include in the top level directory.

This is to facilitate getting the code out of the header files.

No behavioral changes are expected.

BUG=webrtc:6842

Review-Url: https://codereview.webrtc.org/2555993003
Cr-Commit-Position: refs/heads/master@{#15623}
2016-12-15 08:54:15 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value.
2016-12-14 15:58:30 +00:00
audio
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
base
Add WriteIsolatedOutput() functions
2016-12-14 23:03:11 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Disable failing perf test on Android.
2016-12-09 14:15:22 +00:00
common_audio
Fix for left shift of negative value in NetEq.
2016-12-14 10:00:02 +00:00
common_video
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
2016-12-14 10:06:49 +00:00
examples
Adds basic Bluetooth support to AppRTCMobile
2016-12-14 15:37:01 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
2016-12-14 10:06:49 +00:00
modules
Move all codec specific definitions from modules_include
2016-12-15 08:54:15 +00:00
p2p
Fixing possible crash due to RefCountedChannel assignment operator.
2016-12-14 00:38:46 +00:00
pc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sdk
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
2016-12-14 10:06:49 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002
2016-12-14 08:07:44 +00:00
test
Add WriteIsolatedOutput() functions
2016-12-14 23:03:11 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
2016-12-14 10:06:49 +00:00
voice_engine
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +00:00
.gitignore
…
BUILD.gn
Adds basic Bluetooth support to AppRTCMobile
2016-12-14 15:37:01 +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
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +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
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
2016-12-14 10:06:49 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.