Commit Graph

16 Commits

Author SHA1 Message Date
astor@webrtc.org
bd7aeba8fb Expose a set of options to the OveruseDetector supporting experiments
Updated overuse_detector.* to use google style naming convention
Removed OveruseDetector::Reset
Review URL: https://webrtc-codereview.appspot.com/666005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2443 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-26 10:47:04 +00:00
stefan@webrtc.org
dc257b5781 Add option to configure error concealment and disable by default.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2324 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-30 11:25:00 +00:00
stefan@webrtc.org
c8e4886774 Upgrade libvpx to 6b66c01 and enabling temporal denoising.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1921 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-21 16:52:03 +00:00
andrew@webrtc.org
8012474552 Use a const rather than macro for EcDefault.
- This should be a better solution to the build error in
  https://webrtc-codereview.appspot.com/425005
- Ideally all of the similar macros should go away, but one thing at
  a time...

BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/438002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1860 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-08 17:54:24 +00:00
phoglund@webrtc.org
8bfee84144 Initial revision of a ViE fuzz test. The idea is to inject randomized RTP packets and see what the video engine does.
There are some small refactorings in here, but the real focus of this CL is in vie_autotest_rtp_fuzz.cc. This patch is mostly here to get a discussion going.

On my initial test the video engine doesn't recover, at least within 10 seconds of running with untampered packets. Not sure if this is according to specification though.

Ideas:
  - Generate random packets with correct RTP headers to get further into the code.
  - Don't generate fresh random data, but rather corrupt bits here and there in small amounts.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/383001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1714 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 09:32:48 +00:00
pwestin@webrtc.org
5621057956 Removing unused code.
Review URL: https://webrtc-codereview.appspot.com/349008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1442 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 12:45:47 +00:00
henrika@webrtc.org
f75901fa4c Resolves CID 10540: Copy into fixed size buffer (STRING_OVERFLOW).
You might overrun the 32 byte fixed-size string "receiveCodec.plname" by copying "payloadName" without checking the length.
Note: This defect has an elevated risk because the source argument is a parameter of the current function.
Review URL: http://webrtc-codereview.appspot.com/352009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1428 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 08:45:42 +00:00
henrik.lundin@webrtc.org
053c7991e3 Add minimum waiting time to NetEQ metrics
Adding minWaitingTimeMs to ACMNetworkStatistics and to
NetworkStatistics. Also adding unittest.

TEST=audio_coding_unittests

Review URL: http://webrtc-codereview.appspot.com/350006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1408 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 14:16:44 +00:00
henrik.lundin@webrtc.org
d439870473 Adding two new network metrics to NetEQ
Clock-drift (in parts-per-million) and peaky-jitter mode status.
Both metrics are propagated to the VoE API. Tests are added
in the NetEQ unittests, and to some extent in ACM unittests
and VoE tests.

Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.

Note: The file neteq_network_stats.dat in resources must be updated
for the unittests to pass.

Review URL: http://webrtc-codereview.appspot.com/337005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1328 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 13:09:55 +00:00
mikhal@webrtc.org
c00f91d62d Adding BGRA as a video type.
This CL is a prerequisite for the capture module update CL. 
Review URL: http://webrtc-codereview.appspot.com/329021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1317 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 18:49:15 +00:00
stefan@webrtc.org
efd0a48c61 Add error resilient mode options to the VP8 specific VideoCodec struct.
It is useful to disable error resilience when we know we won't decode
with errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1305 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 10:12:35 +00:00
henrik.lundin@webrtc.org
dbba1f969f Packet waiting-time statistics
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.

Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.

Updating common_types.h and VoiceEngine tests to include the
new metrics.

Unit tests are also added for NetEQ and AcmNetEq.

Review URL: http://webrtc-codereview.appspot.com/328011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 15:45:05 +00:00
pwestin@webrtc.org
db221d2b81 Fixes to temporal layers, Henrika please review src/common_types.h
Review URL: http://webrtc-codereview.appspot.com/286001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1091 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 11:31:08 +00:00
henrik.lundin@webrtc.org
df10de4b27 Removing statistics API from NetEQ
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.

Review URL: http://webrtc-codereview.appspot.com/285002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 09:36:23 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine.
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 15:19:55 +00:00
niklase@google.com
470e71d364 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00