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webrtc_m130/webrtc
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deadbeef 0a6c4ca942 Catching more errors when parsing ICE server URLs. 
Every malformed URL should now produce an error message in JS, rather than
silently failing and possibly printing a warning message to the console (and
possibly crashing).

Also added some unit tests, and made "ParseIceServers" public.

BUG=445002

Review URL: https://codereview.webrtc.org/1344143002

Cr-Commit-Position: refs/heads/master@{#10186}
2015-10-06 18:38:33 +00:00
..
audio
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
base
Catching more errors when parsing ICE server URLs.
2015-10-06 18:38:33 +00:00
build
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
call
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Loopback and audio only mode.
2015-10-02 18:45:13 +00:00
libjingle
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
modules
Transport sequence number should be set also for retransmissions.
2015-10-06 13:02:57 +00:00
p2p
Convert uint16_t to int for WebRTC cipher/crypto suite.
2015-10-05 19:43:33 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
test
Remove kSkipFrame usage.
2015-10-06 12:05:03 +00:00
tools
Remove AgcManager.
2015-09-29 22:45:23 +00:00
video
Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
2015-10-06 18:34:14 +00:00
video_engine
Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay):
2015-10-06 11:08:22 +00:00
voice_engine
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
.gitignore
…
audio_receive_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
audio_send_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
BUILD.gn
Update build files to use webrtc_overrides in Chromium instead of overrides.
2015-09-30 10:49:11 +00:00
call.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Unify newapi::RtcpMode and RTCPMethod.
2015-10-02 09:37:18 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Remove kSkipFrame usage.
2015-10-06 12:05:03 +00:00
video_receive_stream.h
Unify newapi::RtcpMode and RTCPMethod.
2015-10-02 09:37:18 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
2015-10-06 18:34:14 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Remove AgcManager.
2015-09-29 22:45:23 +00:00
webrtc.gyp
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +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.