cf1b51b6fb
Moves the display reconfiguration callback into a separate class,
...
so that it can be shared with the cursor monitor when single monitor capturing
is added (https://webrtc-codereview.appspot.com/4679005/ ).
This Cl should have no functionality change.
BUG=2253
R=henrike@webrtc.org , sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 21:59:12 +00:00
f7c6e743b3
Fix deadlock in video_receiver.cc.
...
In webrtc::vcm::VideoReceiver::ResetDecoder(), the lock order is:
1. take _receiveCritSect,
2. take process_crit_sect_
This conflicts with the follow code path:
1. webrtc::vcm::VideoReceiver::Process(), take process_crit_sect_
call -> webrtc::vcm::VideoReceiver::NackList(),
2. with nackStats=kNackKeyFrameRequest, take _receiveCritSect
BUG=2861
TEST=trybots
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5456 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 10:27:51 +00:00
c7c7a531f3
Add Config struct for experimental AGC.
...
Disable in the audio mixer.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5454 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 04:57:25 +00:00
7433a088d2
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
...
We reverted the r5421 to allow us roll webrtc to chrome without any modifications
to libjingle. Since webrtc is rolled with r5444, we can add back the original CL
and changes to libjingle will be upstreamed in the next roll.
TBR=andresp@webrtc.org
> Revert 5421 "Fix deadlock on register/unregister observer while ..."
>
> Failure to compile on Chromium Internal bots, because of API changes.
>
> http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
>
> You need to follow the steps mentioned in
> https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
>
> Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
> as mentioned in the doc.
>
> > Fix deadlock on register/unregister observer while there is a an going callback.
> >
> > BUG=2835
> > R=mallinath@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/7119005
>
> TBR=andresp@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7679004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 00:56:02 +00:00
84eb0e952e
Add clean test to NetEq perf test
...
Add another test to NetEqPerformanceTest with no packet losses or
clock drift. The purpose of this test would be to focus on the
"clean" code path, i.e., the path taken when there are no network
problems. The reason is that this code path is presumably much
lighter in complexity, and regressions could easily drown in the
heavier code involved when combating losses and drift.
BUG=2859
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5452 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-28 21:50:35 +00:00
932b0193e7
VideoCaptureAndroid: stop preview in opposite order of starting.
...
While the SDK documentation doesn't prescribe a required shutdown order, good
hygiene suggests stopping should happen in reverse order of starting. It also
seems to relieve a crash in the system capturer on at least the Galaxy Note 10.
BUG=2793
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:32:05 +00:00
18586d38bc
Revert 5421 "Fix deadlock on register/unregister observer while ..."
...
Failure to compile on Chromium Internal bots, because of API changes.
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
You need to follow the steps mentioned in
https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
as mentioned in the doc.
> Fix deadlock on register/unregister observer while there is a an going callback.
>
> BUG=2835
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7119005
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:00:57 +00:00
a45cac0fb7
Avoid potential dead lock in StreamStatisticianImpl
...
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.
BUG=2856
R=andresp@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 16:22:08 +00:00
5314e85926
Race condition in RTPSender::UpdateRtpStats
...
The ssrc should not be access directly from the ssrc_ field, without
holding the send_critsect_ lock. A better way is to just use the SSRC()
getter method.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 13:20:36 +00:00
2397a17c6b
Fix bug introduced during replace of list wrapper with std equivalents in r5378.
...
R=henrika@webrtc.org , pbos@webrtc.org , henrike@webrtc.org
TBR=henrike@webrtc.org
BUG=2164
Review URL: https://webrtc-codereview.appspot.com/7639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5437 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 12:33:30 +00:00
c00adbed73
Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket
...
StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
be cached while holding lock to avoid race condition.
Also, rtp_callback_ do not need to be called in GetStatistics() at all
BUG=2853
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 10:42:48 +00:00
99eab02fb1
Fix "field '_testNo' is uninitialized" warnings.
...
BUG=2849
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 09:30:35 +00:00
e84978f3d8
Add a Config parameter to AudioProcessing::Create().
...
Also add a parameter-less version; the (int) version is deprecated and
should be removed.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-25 02:09:06 +00:00
871d949299
Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules.
...
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 13:23:49 +00:00
99a8c7e039
Add trace-based delivery filter to BWE test framework.
...
R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 10:00:27 +00:00
8d375c95b7
Fix deadlock on register/unregister observer while there is a an going callback.
...
BUG=2835
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 23:09:25 +00:00
754de528b7
Fix array declarations in aec_rdft.h.
...
Was causing warnings in Chromium such as:
warning C4742: 'rdft_wk2i' has different alignment in
'webrtc\modules\audio_processing\aec\aec_rdft_sse2.c' and
'webrtc\modules\audio_processing\aec\aec_rdft.c': 4 and 16
BUG=chromium:336620
R=cduvivier@google.com
Review URL: https://webrtc-codereview.appspot.com/7489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5419 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 20:55:14 +00:00
0e93257cee
Add callbacks for receive channel RTP statistics
...
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.
TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 10:00:39 +00:00
7dba27c740
Potential dead lock in receive statistics
...
A dead lock could occur if the following to code paths are called
concurrently:
ReceiveStatisticsImpl::IncomingPacket() ->
StreamStatisticianImpl::IncomingPacket()
StreamStatisticianImpl::GetStatistics() ->
ReceiveStatisticsImpl::StatisticsUpdated()
Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().
BUG=2818
R=asapersson@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 16:33:37 +00:00
32c3247418
Fix for libtalkmobile build error
...
bug=b/12549061
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 16:16:58 +00:00
efaeda0c76
Add configuration and test for extended RTCP reference time reports to new video api.
...
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 08:34:49 +00:00
32c26eb90b
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
...
BUG=N/A
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 23:12:51 +00:00
4985927d36
Implement screen enumeration and individual screen capturing for Windows.
...
BUG=2787
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5399 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 17:19:16 +00:00
ead202b973
Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started.
...
BUG=2801
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 23:26:37 +00:00
4ffd9c7423
Add full path to headers
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 09:01:39 +00:00
6a94734d4d
Adds back set_sample_rate_hz() when Init is called in recordings.
...
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.
BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 08:41:09 +00:00
ea9392d5eb
MIPS optimizations for NS audio processing module
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4139006
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 07:22:01 +00:00
fb4e256d49
Fix crash in MouseCursor::CopyOf()
...
This issue was causing test failures with the latest webrtc roll.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7249005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5392 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 04:45:35 +00:00
7a2ca7c621
Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer.
...
This change is also must for rolling webrtc in chrome.
R=jiayl@webrtc.org
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5389 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-15 19:00:13 +00:00
017b619010
Extends the ScreenCapturer interface for individual display screen cast.
...
Real implementations for each platform will be added in future CLs.
BUG=2787
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5386 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 18:26:37 +00:00
39fcfd78ae
Remove empty VideoCodecGeneric struct.
...
Struct was added prematurely and triggers a warning with
-Wextern-c-compat in latest clang.
R=henrika@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/7119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 12:55:59 +00:00
d9faa46d57
Changing to using factory methods for some classes in NetEq
...
In this CL, the Expand, Accelerate and PreemptiveExpand objects are
created using factory methods. The factory methods are injected into
NetEqImpl on creation. This is a step towards implementing a no-decode
operation.
BUG=2776
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5382 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 10:18:45 +00:00
eb31b45aaf
Fix MouseCursorMonitorMac to return correct hotspot position.
...
Previusly (0, 0) was always return as mouse cursor hotspot.
BUG=2779
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 23:25:17 +00:00
3907c2e7e5
Removes the remaining uses of the list wrapper class and the list wrapper class.
...
BUG=2164
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7019007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5378 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 22:41:34 +00:00
79cf3acc79
Removes usage of ListWrapper from several files.
...
BUG=2164
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 15:21:30 +00:00
d0b436a935
Revert "Activate ACM test for Android in modules_tests." (rev5364).
...
TBR=turaj@webrtc.org ,tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 13:15:59 +00:00
8bc4fcfeb6
Temporarily disabling audio processing tests.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6889005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5370 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 09:14:47 +00:00
2c03bf1641
Increasing simulation time for NetEqPerformanceTest
...
This is to get better "signal-to-noise ratio" in the performance bots.
The neteq4-runtime metric is expected to increase by a factor of 10.
BUG=2397
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6989005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5369 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 09:04:23 +00:00
bbd47fc5b5
Enables robust delay validation in AEC delay logging.
...
* Explicitly disabled robust validation in AECM.
* Updated audio_processing_unittests for using robust delay validation in AEC.
* Updated output_data_float.pb (not needed for Android nor fixed point, since AECM is untouched).
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5368 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 08:54:34 +00:00
573a1b45b5
Android: Fixes crash when exiting WebRTCDemo.
...
BUG=2738
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-10 22:58:06 +00:00
7cc64b3747
Activate ACM test for Android in modules_tests.
...
TEST=local on Nexus 7.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5364 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-10 22:35:09 +00:00
a366e810a9
Adding NetEq performance test to webrtc_perf_tests
...
The performance test is based on the neteq4_speed_test application. The
bulk of the test code is extracted into a test class, and included into
the neteq_unittest_tools target. The actual gtest that runs the
performance test is implemented in neteq_performance_unittest.cc, and
built as a part of webrtc_perf_tests.
The old stand-alone test application is now made dependent on the new
test class, to avoid code duplication.
BUG=2397
R=andrew@webrtc.org , kjellander@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5362 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-10 08:24:04 +00:00
fa8d534e09
Delay Estimator: Adds unittests for robust validation.
...
In addition to unittests a cast losing constness was corrected.
The tests added are:
1. Adjusting allowed_offset when robust validation is disabled should have no impact.
2. For noise free signals there should be no difference between robust validation or not.
3. Robust validation acts faster during startup.
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5361 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-10 07:42:07 +00:00
e7ce437333
Fixing lint errors in NetEq4
...
Just taking care of a few old lint errors.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5359 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-09 14:01:55 +00:00
c5aeb2aa15
Make code simpler on VCMEncodedCallback.
...
R=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5358 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-09 08:04:32 +00:00
1df9dc3957
Isolate register post encode callback in video coding module to simplify code and critical sections.
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R=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-09 08:01:57 +00:00
b08a12d6e8
Isolate debug recording from video sender into a thread safe small class.
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R=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08 12:38:22 +00:00
bccd53de57
Delay Estimator: Converts a constant into a configurable parameter.
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The parameter is used in the robust validation scheme, which will be turned on in a separate CL.
* Setter and getter for allowed delay offset.
* Updated unittests.
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5351 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08 08:18:15 +00:00
d335094852
Init to 16 kHz in the fixed-point profile.
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Fixes modules_unittests for fixed-point builds (Android).
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/6709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5349 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 18:57:10 +00:00
b6541ca3a1
Ensure capture_levels_ is sized correctly at init time.
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Fixes failing voe_auto_test and audioproc_perf.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/6699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5348 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 18:36:10 +00:00
