platform-external-webrtc

Files

andrew@webrtc.org d59359af4d Remove 44.1 kHz workaround from the iOS AudioDevice.

Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.

This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.

BUG=1395
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-04-22 18:07:49 +00:00

build

Set include_internal_video_capture=1 for video_capture_tests

2014-04-16 12:59:49 +00:00

common_audio

common_audio: VADFree() now returns void

2014-04-22 04:45:35 +00:00

common_video

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

examples

VideoCaptureAndroid: stop referencing ViERenderer

2014-04-08 22:55:07 +00:00

modules

Remove 44.1 kHz workaround from the iOS AudioDevice.

2014-04-22 18:07:49 +00:00

system_wrappers

Add Chromium's ScopedVector.

2014-04-18 21:20:54 +00:00

test

Remove use of tmpnam.

2014-04-16 08:04:26 +00:00

tools

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

video

Let A/V sync test use default AudioCoding module

2014-04-22 08:36:33 +00:00

video_engine

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

voice_engine

Revert "Stop using ACM factory in VoiceEngine"

2014-04-22 09:44:00 +00:00

.gitignore

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

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

call.h

Remove internal codecs from VideoSendStream.

2014-03-19 08:43:57 +00:00

common_types.h

Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-04-17 10:45:01 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

engine_configurations.h

Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-04-14 14:12:50 +00:00

experiments.h

Adding API for setting bandwidth estimation configurations.

2014-03-25 10:37:31 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

OWNERS

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

PATENTS

Move src/ -> webrtc/

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

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +00:00

README.chromium

Move src/ -> webrtc/

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

supplement.gypi

Removes script for generating supplement.gypi also adds git ignore for tools/gn.

2014-01-21 15:54:56 +00:00

transport.h

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

2013-11-20 12:17:04 +00:00

typedefs.h

Fix ARM64 detection.

2014-04-01 01:19:08 +00:00

video_engine_tests.isolate

Merge metrics_unittests into video_engine_tests.

2013-12-13 14:31:47 +00:00

video_receive_stream.h

Make VideoReceiveStream::GetStats() const.

2014-02-07 15:32:45 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Simplify pacer interface.

2014-03-19 10:59:52 +00:00

webrtc_examples.gyp

Make WebRTC Android examples build without sourcing envsetup.sh

2014-04-15 08:35:00 +00:00

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +00:00

webrtc_tests.gypi

Adding NetEq performance test to webrtc_perf_tests

2014-01-10 08:24:04 +00:00

webrtc.gyp

Integrate fake_network_pipe into direct_transport.

2013-12-18 20:28:25 +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.