platform-external-webrtc

Files

mcasas@webrtc.org 6e2d012b58 Add support for YUV4MPEG file reading to tools files.

This CL adds support for reading .y4m files to the infra in
video_quality_analysis.cc, adding new functions 
ExtractFrameFromYuvFile() and ExtractFrameFromY4mFile(),
instad of the previous ExtractFrameFromI420(). The decision
as to which one to use is taken from the file extension,
if it is .y4m then is considered a YUV4MPEG file, otherwise
is taken as a raw .yuv file.

It also removes the pseudo duplicated function 
GetNextI420Frame(), that is used from psnr_ssim_analyzer.c,
and adds support for y4m files there.

Tested/validated via local compile-run.

YUV4MPEG is a trivial container with a file header
and a per-frame header, see [1]

[1]
http://wiki.multimedia.cx/index.php?title=YUV4MPEG2

BUG=https://code.google.com/p/chromium/issues/detail?id=343504



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5702 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-03-14 12:45:45 +00:00

build

Enable VS2013 for Windows compilation by default.

2014-03-12 14:41:58 +00:00

common_audio

Add a float interface to PushSincResampler.

2014-03-10 18:51:42 +00:00

common_video

Add an AlignedFreeDeleter and remove scoped_ptr_malloc.

2014-02-20 21:08:36 +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

Fix a bug where network freeze during CNG causes delay

2014-03-14 12:40:05 +00:00

system_wrappers

Remove std:: prefixes from C functions in webrtc/.

2014-03-07 15:23:34 +00:00

test

Implement minimum transmit bitrate.

2014-03-13 12:52:27 +00:00

tools

Add support for YUV4MPEG file reading to tools files.

2014-03-14 12:45:45 +00:00

video

Stopping network threads before tearing down test

2014-03-13 15:39:27 +00:00

video_engine

Implement minimum transmit bitrate.

2014-03-13 12:52:27 +00:00

voice_engine

Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now.

2014-03-11 16:19:56 +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

Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).

2014-02-19 11:59:02 +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

Remove external encryption API for VoE.

2014-02-18 11:27:22 +00:00

experiments.h

Missing include in experiments.h

2014-02-25 09:17:43 +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

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

Moves WEBRTC_POSIX define from header file to gyp-settings.

2014-03-07 15:30:21 +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

Re-landing "Routing SuspendChange to VideoSendStream::Stats"

2014-03-13 13:31:21 +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.