platform-external-webrtc

Files

hbos e448dd5355 RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector

and requestsSent is updated.

Before:
  requestsSent = total ping requests
Now
  requestsSent = pings sent before first response
  consentRequestsSent = pings after first response

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-consentrequestssent

BUG=chromium:633550

Review-Url: https://codereview.webrtc.org/2558293002
Cr-Commit-Position: refs/heads/master@{#15541}

2016-12-12 09:22:59 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector

2016-12-12 09:22:59 +00:00

audio

Refactor webrtc/{api,audio} and modules/audio_coding for GN check

2016-12-07 16:23:35 +00:00

base

Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves

2016-12-12 02:42:14 +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

Split targets mixing .c and .cc sources.

2016-12-06 06:47:52 +00:00

common_video

Sanity check parsed QP values from H264 bitstream

2016-12-02 09:29:53 +00:00

examples

Replace VideoCaptureDataCallback by VideoSinkInterface.

2016-12-12 08:23:05 +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

Replace VideoCaptureDataCallback by VideoSinkInterface.

2016-12-12 08:23:05 +00:00

modules

Replace VideoCaptureDataCallback by VideoSinkInterface.

2016-12-12 08:23:05 +00:00

p2p

Implement the "needs-ice-restart" logic for SetConfiguration.

2016-12-10 21:15:39 +00:00

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

sdk

Modify JavaToStdString to allow ISO-8859-1 encoded strings.

2016-12-09 14:13:48 +00:00

stats

Collecting RTCIceCandidatePairStats.transport_id and improved unittests.

2016-11-30 09:50:36 +00:00

system_wrappers

Add comment to metrics.h

2016-12-06 11:59:08 +00:00

test

Replace VideoCaptureDataCallback by VideoSinkInterface.

2016-12-12 08:23:05 +00:00

tools

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

video

Only store sequence numbers for media stream in FlexFEC end-to-end test.

2016-12-09 14:51:43 +00:00

voice_engine

Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h

2016-12-05 10:22:18 +00:00

.gitignore

…

BUILD.gn

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +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

…

transport.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

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

Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )

2016-11-11 11:55:19 +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

voice_engine_configurations.h

Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h

2016-12-05 10:22:18 +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.