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nisse 38cc1d6b31 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. 
This avoids redoing RTP header parsing already done in Call, for video.

The next step is to convert other types of receive streams, i.e.,
audio and flexfec, to use a compatible OnRtpPacket method. We can then
introduce a shared base interface, and simplify media-independent
receive processing in Call.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2681673004
Cr-Commit-Position: refs/heads/master@{#16583}
2017-02-13 13:59:46 +00:00
..
api
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
audio
Clean out platform specific things from voice_engine_defines.h.
2017-02-13 12:42:52 +00:00
base
Delete unused classes AutoDetectProxy and SslSocketFactory.
2017-02-13 12:33:28 +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
Add support for swapping feeds in Android AppRTCMobile.
2017-02-13 13:58:27 +00:00
logging
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
media
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
modules
Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
2017-02-13 13:59:46 +00:00
p2p
Rename "PacketTransportInterface" to "PacketTransportInternal".
2017-02-10 19:31:50 +00:00
pc
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
sdk
Add ifdef protection for iOS-only headers
2017-02-13 13:09:01 +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
Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
2017-02-13 13:59:46 +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
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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.