webrtc_m130

History

henrika 324d9c9a86 Avoids error message about unknown selected data source for Port iPhone Microphone

TBR=tkchin
BUG=webrtc:4845
TEST=modules_unittests

Review URL: https://codereview.webrtc.org/1237233003 .

Cr-Commit-Position: refs/heads/master@{#9602}

2015-07-20 11:09:34 +00:00

base

Ensuring that UDP TURN servers are always used as STUN servers.

2015-07-16 17:22:28 +00:00

build

Make .gni less sensitive to type of arm_use_neon flags

2015-07-09 16:45:09 +00:00

common_audio

Remove a cast again, after it was shown to worsen Windows perf.

2015-06-29 22:16:48 +00:00

common_video

Re-land "Convert native handles to buffers before encoding."

2015-06-05 09:08:12 +00:00

examples/android/media_demo

VoE: Remove unused interfaces

2015-05-26 08:25:00 +00:00

libjingle

Remove webrtc/libjingle/{examples,session}.

2015-06-12 11:13:46 +00:00

modules

Avoids error message about unknown selected data source for Port iPhone Microphone

2015-07-20 11:09:34 +00:00

overrides

Attempt at fixing error on the Chrome Windows FYI bots.

2015-05-25 19:22:23 +00:00

p2p

Fix an NPE when creating TurnPort with a NULL socket.

2015-07-17 23:22:02 +00:00

sound

Remove henrike@ from OWNERS

2015-04-01 15:08:49 +00:00

system_wrappers

Fix some clang warnings with -Wmissing-braces in WebRTC.

2015-07-02 00:52:18 +00:00

test

Add pcap support to bwe tools. Allow filtering on SSRCs.

2015-07-17 12:27:27 +00:00

tools

Remove empty-string comparisons.

2015-07-09 14:48:27 +00:00

video

Define Stream base classes

2015-07-16 07:30:20 +00:00

video_engine

Remove unused metric in overuse detector.

2015-07-16 06:08:20 +00:00

voice_engine

Remove ResetStatistics from RTP feedback.

2015-07-07 15:32:56 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

audio_receive_stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

audio_send_stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

BUILD.gn

iOS HW H264 support.

2015-06-29 21:35:08 +00:00

call.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

codereview.settings

Update deeper codereview.settings files to match the root.

2015-06-24 19:40:39 +00:00

common_types.cc

Parsing of transport wide sequence number rtp extension header.

2015-03-17 14:33:46 +00:00

common_types.h

Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545.

2015-07-10 09:28:46 +00:00

common.gyp

Fix style violations in common_types.h and config.h

2015-02-26 14:01:28 +00:00

common.h

Make Config::default_value leak instead of having an exit-time destructor.

2015-05-23 00:50:33 +00:00

config.cc

Add AudioReceiveStream to Call API.

2015-04-29 13:24:10 +00:00

config.h

Add options for NetEq fast accelerate mode through libjingle

2015-06-01 08:29:55 +00:00

engine_configurations.h

Remove VideoEngine interfaces.

2015-05-12 14:51:08 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

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

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

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

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

supplement.gypi

Move target_subarch from gyp_webrtc to supplement.gypi

2015-04-16 07:24:23 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +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

Re-land "Convert native handles to buffers before encoding."

2015-06-05 09:08:12 +00:00

video_receive_stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

video_renderer.h

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

video_send_stream.h

Define Stream base classes

2015-07-16 07:30:20 +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

Move frame input (ViECapturer) to webrtc/video/.

2015-06-26 04:58:23 +00:00

webrtc.gyp

Define Stream base classes

2015-07-16 07:30:20 +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.