20 Commits

Author SHA1 Message Date
phoglund@webrtc.org
69d46b4821 Added basic fuzzer for new API and made both work.
Added a nice mode, cleaned up.

BUG=

Review URL: https://webrtc-codereview.appspot.com/832004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2807 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 07:44:02 +00:00
kjellander@webrtc.org
d0b31b2e03 Test page for Apprtc in an iframe
BUG=None
TEST=Browsed to the page and started video chat.

Review URL: https://webrtc-codereview.appspot.com/811005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2788 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-19 13:46:19 +00:00
phoglund@webrtc.org
daf32a4cb9 Adapted fuzzer for launch.
BUG=

Review URL: https://webrtc-codereview.appspot.com/813006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2787 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-19 13:44:50 +00:00
phoglund@webrtc.org
4c6c11553c Initial version of a peerconnection fuzzer.
The fuzzer can either pass random utf-8 into the SDP parser or swap lines in the generated SDP offer or answer. I've tried to implement the fuzzer so that all random choices are coded into the javascript page so that the sole source of randomness is in the fuzzer program. I initially tried to load stored sample SDP offers and fuzz them in the fuzzer program, but it didn't work since the SDP message seems to contain some magic checksum that causes the parser to choke quickly.

There's a lot of ideas for follow up patches:
- Fuzz ALL input parameters to ALL functions, not just SDP
- Swap letters/words in SDP messages
- Insert random location.reload() anywhere in the call sequence
- Swap lines in the call sequence itself

BUG=

Review URL: https://webrtc-codereview.appspot.com/784004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2772 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-14 10:06:57 +00:00
andrew@webrtc.org
fa418ac0af Consolidate common_video targets to improve gyp run time.
Not sure if this change is measurable; perhaps a 1% savings.

BUG=webrtc:34

Review URL: https://webrtc-codereview.appspot.com/785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2732 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-11 01:34:21 +00:00
vspasova@webrtc.org
f61dc9be41 Moving frame_analyzer and rgba_to_i420_converter to src/tools.
It might be useful to have these under src/tools as this way they will automatically sync in Chrome.

BUG=

Review URL: https://webrtc-codereview.appspot.com/740004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2653 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-22 08:12:00 +00:00
vspasova@webrtc.org
fd80070aa7 A tool to do PSNR and SSIM analysis over frames.
This is a very simple tool which takes a test and reference YUV files and compares the frames in them.
The interesting part is that the test video is the video formed by the frames captured from the browser
(WebRTC output). The test video frames has been decoded and the correspondence between evrey frame in the
test video and the decoded barcode, i.e. the frame number in the refrence video has been written to a stats
file in the form frame_xxxx yyyy, where xxxx is the number of the frame in the test video and yyyy is the
number of the frame in the reference video.

We can have jumping over frames or duplicate frames in the test video, as well as incorrectly decoded barcodes.
The tool takes care of these cases.

I haven't used the video_metrics.h because the functions in there seem to do much more than we need and not to
do things that I needed.

The tool may need to be changed so that it could produce output which in turn will be used by PythonCharts or
other chart-drawing tool or library that we decide to use.

BUG=
TEST=
./out/Debug/frame_analyzer --reference_file=reference.yuv --test_file=test.yuv --stats_file=stats.txt --width=352 --height=288

Review URL: https://webrtc-codereview.appspot.com/701005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2620 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-16 14:07:02 +00:00
vspasova@webrtc.org
8e221ee102 Making the RGBA to I420 tool more useful. Did the following changes:
- Made the output file to open in write mode instead of append mode.
- Now the tool deletes the RGBA frames after conversion.
- Other minor cleanup work.

BUG=

TEST=
rgba_to_i420_converter --frames_dir=<directory_to_rgba_frames> --output_file=<output_yuv_file> --width=<width_of_input_frames> --height=<height_of_input_frames>

Review URL: https://webrtc-codereview.appspot.com/728004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2611 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-15 07:42:00 +00:00
phoglund@webrtc.org
48cf686933 Removed v4l2_file_player code, which is checked into the signal repo.
BUG=

Review URL: https://webrtc-codereview.appspot.com/718007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2581 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09 14:57:58 +00:00
phoglund@webrtc.org
54e22eb977 Made it possible to run video_capture tests on mac.
Abstracted out a suitable main from vie_auto_test and put it into testsupport.
Cleaned up unused vie_auto_test mac code.

BUG=

Review URL: https://webrtc-codereview.appspot.com/723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2572 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-08 08:27:46 +00:00
andrew@webrtc.org
d7a71d0719 Prepare to roll Chromium to 149181.
- This roll brings in VS2010 by default. The buildbots
  need updating (issue710).
- We'll roll to 149181 later (past current Canary) to fix
  a Mac gyp issue:
  https://chromiumcodereview.appspot.com/10824105
- Chromium is now using a later libvpx than us. We should
  investigate rolling our standalone build.
- Fix set-but-unused-warning
- Fix -Wunused-private-field warnings on Mac.

TBR=kjellander@webrtc.org
BUG=issue709,issue710
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/709007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-01 01:40:02 +00:00
andrew@webrtc.org
c1354bd768 Make handling of libyuv more flexible.
- Use gyp variable for libyuv path.
- Rename internal libyuv.h to webrtc_libyuv.h to avoid conflicts.
- Update affected includes.

BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/711004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2534 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-27 18:21:16 +00:00
andrew@webrtc.org
d41f59a23f Fix Mac-gcc warnings.
Resolves:
- warning: allocating zero-element array
- warning: suggest a space before ‘;’ or explicit braces around empty
  body in ‘for’ statement

BUG=none
TEST=build on Mac-gcc, trybots

Review URL: https://webrtc-codereview.appspot.com/675006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2519 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-16 17:05:47 +00:00
phoglund@webrtc.org
2eefb2242f Improved fuzzer. It will now throw in additional refreshes, which is known to mess with lifetime assumptions.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/679008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2492 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-04 12:29:09 +00:00
phoglund@webrtc.org
ef8ca6a801 Wrote ClusterFuzz test for WebRTC GetUserMedia.
This initial test is very simple since we are just releasing GetUserMedia in the next release.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/639006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2476 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-02 11:39:22 +00:00
vspasova@webrtc.org
b358bd8f87 A command-line tool based on libyuv to convert a set of RGBA files to a YUV video.
BUG=
TEST=
tgbra_to_i420_converter --frames_dir=<directory_to_rgba_frames> --output_file=<output_yuv_file> --width=<width_of_input_frames> --height=<height_of_input_frames>

<output_yuv_file> should be an empty file because we open it in append mode

Review URL: https://webrtc-codereview.appspot.com/673006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2475 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-02 07:43:30 +00:00
leozwang@webrtc.org
fb59442c40 Change file path to make it work on android
BUG=
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/672007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2471 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-29 18:28:12 +00:00
andrew@webrtc.org
7a281a5634 Fix Android build after test/ -> src/test/
TBR=leozwang@webrtc.org
BUG=none
TEST=Android trybot

Review URL: https://webrtc-codereview.appspot.com/677006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2447 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-27 03:22:37 +00:00
andrew@webrtc.org
81cf5e4752 Move test to src/test.
- Refer to top-level directories by <(DEPTH), e.g. <(DEPTH)/testing.
- Remove now unneeded third_party_root.

TBR=henrike@webrtc.org
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/669007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2446 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-27 01:41:54 +00:00
phoglund@webrtc.org
1ad477de3e Added a bit flip fuzz test to the voice engine.
Extracted encryption classes to a new test library.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/564009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2256 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-18 08:02:37 +00:00