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Author SHA1 Message Date
wu@webrtc.org
07a6fbe83d Update talk to 56092586. 
R=jiayl@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 18:41:34 +00:00
kjellander@webrtc.org
3779c1cb0a Fix invalid .sha1 files for audio_coding 
It seems like multiple runs of the upload_to_google_storage.py
script created .sha1.sha1 files that sneaked in with
https://code.google.com/p/webrtc/source/detail?r=5076

This caused the wrong files getting downloaded during sync.
This affected the modules_unittests and the neteq_unittests
which started failing (due to wrong version of the resource files).

TEST=trybots passing
BUG=2294
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 14:54:47 +00:00
kjellander@webrtc.org
80174583bd Replace old resources download script with depot_tools 
With help from hinoka@, we're now using a more efficient approach
to download only the files that have changed from Google Storge.

When uploading new resource files, use
upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename
which of course requires gsutil authentication setup.

NOTICE: Before deploying this, svn:ignore should be removed for
the resources folder, or the bots will run into problems with a
non-versioned file being found in the checkout during sync (as
this CL adds resources to version control).

All developers will also need to be informed to wipe their local
resources dir to avoid getting an error during checkout due to the
already existing non-versioned resources directory.

BUG=2294
TEST=locally running gclient runhooks
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2095004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 12:07:57 +00:00
kjellander@webrtc.org
a452fc29e6 Remove resources/ svn:ignore to prepare for updated resource handling 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 12:07:09 +00:00
kjellander@webrtc.org
58bcdeee2c Roll chromium_revision 229708:231713 
Recent changes in how the build dir is used for bots
(see https://codereview.chromium.org/38873003 for details)
requires us to roll to a more recent version
of Chromium to get our android_apk trybot back into
a working state.

This roll needs to be landed at the same time as the
client.webrtc and tryserver.webrtc masters are updated
with the changes in https://codereview.chromium.org/53283002

TEST=trybots passing (except the iOS ones since they require
the above change to be applied to be able to compile)
BUG=2560
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 09:40:03 +00:00
asapersson@webrtc.org
766154aa1d Removed unused code. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 08:35:50 +00:00
kjellander@webrtc.org
e2df8b7f01 Make video quality analysis unittests print to log instead of stdout. 
I think it's best to avoid printing these perf numbers since
when we turn on perf measurements for Android, it will be for
all tests as far as I understand it works today.

TEST=trybots passing tools_unittests
BUG=none
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-03 18:34:51 +00:00
sheu@chromium.org
5dd2ecb32d Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.

TBR=niklas.emblom@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/3269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 23:41:04 +00:00
sheu@chromium.org
74e6e8458e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:48:16 +00:00
sheu@chromium.org
d705649edf Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.

Revert while build breakage is fixed.

BUG=None
TBR=niklas.emblom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:20:15 +00:00
sheu@chromium.org
1a4ed0d70c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 20:32:28 +00:00
wu@webrtc.org
de305014c6 Update talk to 55906045. 
Review URL: https://webrtc-codereview.appspot.com/3159005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:40:38 +00:00
turaj@webrtc.org
58cd31665c Address Clag Analyzer issues. 
Following are the issues related to NetEq 4, discovered by Clang Analyzer.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath
Valid; perhaps unlikely, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath
Valid, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath
Valid; Addressed

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html
Not valid;

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath
Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath
Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath
Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath
Valid; addressed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:15:55 +00:00
asapersson@webrtc.org
7d6bd22019 Propagate estimated RTT from receivers to rtt observer. 
BUG=1613
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 12:14:34 +00:00
sprang@webrtc.org
da2c37b759 Video bandwidth not reported correctly 
ViEChannel::GetBandwidthUsage fails to aggregate video_bitrate_sent in
the same way as the total, fec and nack.

BUG=2579
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 09:49:03 +00:00
sergeyu@chromium.org
773e72797f Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc 
Chromium issue:
https://code.google.com/p/chromium/issues/detail?id=310146

BUG=2551
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2759004

Patch from Daniel Nicoara <dnicoara@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 01:51:21 +00:00
wu@webrtc.org
de748c806c Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. 
TEST=build
R=andrew@webrtc.org, fischman@webrtc.org
TBR=andrew

Review URL: https://webrtc-codereview.appspot.com/3149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 20:43:27 +00:00
solenberg@webrtc.org
dce70ccb0b Add delay limit to ChokeFilter. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 19:18:07 +00:00
wu@webrtc.org
f424cb8e13 Update talk to 55863981. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/3089006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 17:57:33 +00:00
solenberg@webrtc.org
d6e46638ec Logging for BWE test framework. 
BUG=
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 16:06:26 +00:00
wu@webrtc.org
cecfd1832d Update talk to 55821645. 
TEST=try bots
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 05:18:12 +00:00
wu@webrtc.org
ec4cccc6b6 Update libyuv to 832. 
R=fbarchard@google.com, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 21:02:20 +00:00
pbos@webrtc.org
47ebbaddbb Make video/ only depend on video_engine_core. 
Fixes Android/Chromium build error. Previous dependencies included
VideoEngine tests that couldn't build on this configuration.

BUG=2535
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 13:11:56 +00:00
pbos@webrtc.org
def22b455b Stop DirectTransports in VideoSendStreamTests. 
Prevents racy packet delivery during or after Call destruction.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3099005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 10:12:10 +00:00
turaj@webrtc.org
55e1723713 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. 
BUG=2515
TEST=reproduced locally on linux and verified the fix resolves the issue.
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 04:40:09 +00:00
fischman@webrtc.org
9ca93a8b8e Explicitly @synthesize ObjC @properties 
This is required after https://code.google.com/p/gyp/source/detail?r=1768
turned on -Wobjc-missing-property-synthesis for ninja builds (until then it
was only enabled for xcode builds) to allow chromium_deps to roll in
webrtc/DEPS.

BUG=2560
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 00:14:15 +00:00
mikhal@webrtc.org
0aeb22e32c Adding tl0idx consideration for continuity 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 22:26:14 +00:00
pbos@webrtc.org
0803c03f9a Fix build/isolate.gypi path in webrtc_tests.gypi. 
BUG=2535
R=kjellander@webrtc.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 18:10:29 +00:00
fischman@webrtc.org
b7a171825b Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 17:36:59 +00:00
pbos@webrtc.org
16e03b7bd8 Separate Call API/build files from video_engine/. 
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 16:32:01 +00:00
pbos@webrtc.org
850bcbe855 Remove frame_callback.h include in webrtcvie.h. 
This file is about to be moved and it's not really needed. The class
I420FrameCallback is forward declared inside vie_image_process.h and
only used in talk/ for a no-op implementation that doesn't access the
pointer.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 15:41:17 +00:00
henrik.lundin@webrtc.org
1a3a6e5340 Removing the threshold from the auto-mute APIs 
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.

BUG=2436
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 10:16:14 +00:00
sprang@webrtc.org
fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. 
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.

BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 09:21:07 +00:00
wu@webrtc.org
97077a3ab2 Update libjingle to 55618622. 
Update libyuv to r826.

TEST=try bots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 21:18:33 +00:00
fischman@webrtc.org
728bc0fa4c Add qiang.lu@intel.com to WATCHLISTS. 
(patch from http://review.webrtc.org/2859004/)

TBR=qiang.lu@intel.com

Review URL: https://webrtc-codereview.appspot.com/2989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5037 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 19:20:02 +00:00
xians@webrtc.org
c94abd313e Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 18:15:09 +00:00
wu@webrtc.org
e4e5683b41 Clean up tsan suppression file: 
1) remove suppressions that are already fixed.
2) merge duplicated suppressions.

TBR=mallinath
TEST=tsan try bot
BUG=1205,2078,2080

Review URL: https://webrtc-codereview.appspot.com/2949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 16:29:33 +00:00
xians@webrtc.org
0729460acb Added a "interleaved_" flag to webrtc::AudioFrame. 
And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.

BUG=
TEST=compile
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 12:50:46 +00:00
vikasmarwaha@webrtc.org
442c5e47cd Update adapter.js to use TURN transport parameters for FF version 27 & above. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 20:31:57 +00:00
vikasmarwaha@webrtc.org
d674a566d3 Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511. 
R=dutton@google.com, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 19:38:47 +00:00
andrew@webrtc.org
b3731da68f Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. 
Will fix a redefinition error in Chromium against webrtc head.

TESTED=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 15:16:53 +00:00
henrik.lundin@webrtc.org
b56d0e383e Change the low-bitrate handling in BitrateControllerImpl 
Changing to using strategy classes rather than having two different
derived classes of BitrateControllerImpl. This enables run-time switching
of the strategy, which is now possible through a new API. The reason is
that it must fit the current design of ViE.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 09:24:06 +00:00
fischman@webrtc.org
37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. 
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 23:59:45 +00:00
wu@webrtc.org
d371a29227 Fix tsan failures for libjingle_unittest. 
1) Change AsyncSocket's SignalReadEvent and SignalWriteEvent's thread mode to multi_threaded_local as they can be accessed from different threads.
2) Protect NATServer::TransEntry::whitelist.
3) Protect PhysicalSocket:error_.

Detail failures can be seen from issue 2080, comment #5.

TBR=fischman@webrtc.org

RISK=P1
TEST=try bots and tsanv2
BUG=2080

Review URL: https://webrtc-codereview.appspot.com/2669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 23:56:09 +00:00
andrew@webrtc.org
d1bcf1180a Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. 
Works around a multiple definition error from webrtc and libjingle.

Corresponds to the libjingle change here:
https://critique.corp.google.com/#review/55489575-p10

TESTED=trybots
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 19:11:32 +00:00
andrew@webrtc.org
22858d4785 Add an extended filter option to audioproc. 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 14:07:17 +00:00
asapersson@webrtc.org
042e91c2b2 Fix for incorrect RTT estimation. A too low RTT value could be estimated. 
R=andrew@webrtc.org, holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 13:58:31 +00:00
henrik.lundin@webrtc.org
ba975e2078 Porting auto mute to new ViE API 
This CL also includes tests for the auto mute function. A few minor lint
warnings were fixed too. Note that the auto mute function is still work
in progress.

The callback ViEEncoderObserver::VideoAutoMuted was not ported from the
old API. This is TBD; see issue 2457.

BUG=2436
R=holmer@google.com, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 11:04:57 +00:00
tina.legrand@webrtc.org
886aef09a8 Fixing broken tests in voe_auto_test extended 
This CL fixes the problem with voe_auto_test extended-codec test, as well as
extended-file test. First problem was that Opus was not added as a special case, like the other codecs, and the second problem was that the tests were not updated when test files were moved to the resources catalogue.

There are still some tests that fails. Here is a list of all extended tests and their status:

Base: fails - the reason seem to be that external transport has been removed.
CallReport: passes
Codec: passes (with this CL)
DTMF: passes
Encryption: fails or is dissabled?
VoEExternalMedia: passes
File: passes (with this CL)
Hardware: passes
NetEqStats: empty?
Network: passes
RTP_RTCP: fails
VideoSync: fails
VolumeControl: passes

BUG=issue2234
R=andrew@webrtc.org, henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2023004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 10:39:56 +00:00
wu@webrtc.org
8804a29951 Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread. 
TEST=try bots
BUG=1205
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 23:09:20 +00:00