platform-external-webrtc

Files

henrik.lundin@webrtc.org 75642fcd9a Implementing replacement audio support in neteq_rtpplay

This CL makes it possible to replace the payload in an RTP stream
with audio from another (PCM) file. The new payload will be encoded as
PCM16b. The RTP headers will be updated to reflect this change of
payload type.

BUG=2834
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5490 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-02-05 08:49:13 +00:00

build

Add gyp_webrtc script to generate projects.

2014-01-31 09:34:51 +00:00

common_audio

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00

common_video

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

examples

Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.

2014-01-24 18:42:12 +00:00

modules

Implementing replacement audio support in neteq_rtpplay

2014-02-05 08:49:13 +00:00

system_wrappers

Moves the display reconfiguration callback into a separate class,

2014-01-29 21:59:12 +00:00

test

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +00:00

tools

Now printing less output from compare_videos.py.

2014-01-07 17:59:30 +00:00

video

Add configuration for cpu overuse detection to video send stream.

2014-01-31 10:05:07 +00:00

video_engine

Connect webrtc::Config to WrappingBitrateEstimator

2014-01-29 08:47:15 +00:00

voice_engine

Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents"

2014-02-04 10:45:14 +00:00

.gitignore

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

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

call.h

Add configuration for cpu overuse detection to video send stream.

2014-01-31 10:05:07 +00:00

common_types.h

Remove empty VideoCodecGeneric struct.

2014-01-14 12:55:59 +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 RTX in VideoReceiveStream.

2014-01-24 09:30:53 +00:00

engine_configurations.h

Adds a new voice engine warning for the typing noise off state.

2013-09-17 18:09:20 +00:00

experiments.h

Connect webrtc::Config to WrappingBitrateEstimator

2014-01-29 08:47:15 +00:00

frame_callback.h

Implement and test EncodedImageCallback in new ViE API.

2013-11-26 11:41:59 +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

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

Correctly define OVERRIDE when building with g++ 4.7 and C++11 support

2013-12-10 19:20:46 +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

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +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

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +00:00

webrtc_examples.gyp

Android example apps: fixes issue where useful failure information was suppressed.

2014-01-21 19:03:51 +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.