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Author SHA1 Message Date
phoglund@webrtc.org
fa042ca15d Making WebRTC able to play and record audio to files for tests. 
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 09:57:23 +00:00
henrik.lundin@webrtc.org
c726b1fc33 Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" 
BUG=3469
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 08:35:53 +00:00
bjornv@webrtc.org
18026abd82 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 
This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 06:53:20 +00:00
bjornv@webrtc.org
782978cfcb common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix 
This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 06:39:03 +00:00
bjornv@webrtc.org
3f83072c26 modules/audio_processing: Adds a config for reported delays 
There are platforms and devices where the reported delays are untrusted and we currently solve that with an extended filter length and a slightly more conservative delay handling.
With this change we give the user the possibility to turn off reported system delay values completely.

- Includes new unit tests.

TESTED=trybots and manual testing
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 04:48:11 +00:00
henrik.lundin@webrtc.org
fd59c39caa Delete last file in neteq4 folder 
The .isolate file can now be safely removed, since issue 3462 is
resolved.

BUG=2996
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 20:26:27 +00:00
andrew@webrtc.org
919914d71b MIPS optimizations for ISAC (patch #1) 
Implemented functions:
    - WebRtcIsacfix_AutocorrMIPS
    - WebRtcIsacfix_FilterArLoop
    - WebRtcIsacfix_FilterMaLoopMIPS
    - WebRtcIsacfix_AllpassFilter2FixDec16MIPS (only MIPS DSP)
    - WebRtcIsacfix_PitchFilterCore (only MIPS DSPR2)

Gain achieved: from aprox. 15% (MIPS32) up to aprox. 40% (MIPS DSPR2)

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17559005

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 18:13:15 +00:00
kwiberg@webrtc.org
12cd443752 Noise suppression: Change signature to work on floats instead of ints 
Internally, it already worked on floats. This patch just changes the
signature of a bunch of functions so that floats can be passed
directly from the new and improved AudioBuffer without converting the
data to int and back again first.

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the noise suppressor comes immediately after
the echo canceller, which also works on floats. If I truncate to
integers between the two steps, ApmTest.Process doesn't complain, but
of course that's exactly the sort of thing the float conversion is
supposed to let us avoid...)

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 11:13:09 +00:00
kjellander@webrtc.org
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate 
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:42:53 +00:00
henrik.lundin@webrtc.org
620048172c Create a joint encoder/decoder wrapper for iSAC in ACM 
This CL extends the ACMISAC wrapper class to inherit from AudioDecoder
as well (the type of object that NetEq uses). The class has it's own
lock protecting the iSAC instance. This way, we can remove the
neteq_decode_lock_ (a.k.a. decoder_lock_) in a later CL.

The old AcmAudioDecoderIsac class is deleted.

R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 18:39:00 +00:00
henrik.lundin@webrtc.org
a90abdef62 Add thread annotations to AcmReceiver 
This change adds thread annotations to AcmReceiver. These are the
annotations that could be added without changing acquiring the locks in
more locations, or changing the lock structure.

BUG=3401
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 18:35:11 +00:00
bjornv@webrtc.org
ed7edb8e89 Enables DelayCorrection tests 
The fix has been done elsewhere and the test pass.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15679007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 10:02:05 +00:00
henrik.lundin@webrtc.org
a1a2c0c190 Multi-threaded unit test for Audio Coding Module using iSAC 
This test extends AudioCodingModuleTest and AudioCodingModuleMtTest
to using iSAC as codec.

R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 09:37:17 +00:00
bjornv@webrtc.org
cb0ea43e57 audio_processing: Forces extended filter to be used in splitting filter test. 
The behavior differ between "normal" and "extended" modes when using AEC. In the extended filter mode nothing is processed until we have received a farend frame. This is exactly what is needed in this part of the splitting filter test.
On Android, we do not use the normal mode, which made the test to fail.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 08:21:52 +00:00
henrik.lundin@webrtc.org
9c55f0f957 Rename neteq4 folder to neteq 
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.

This CL effectively reverts r6257 "Rename neteq4 folder to neteq".

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 08:10:28 +00:00
henrik.lundin@webrtc.org
9221ab420d Re-enable AudioCodingModuleMtTest again 
Increase timeout and decrease test length.

BUG=3426
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15679006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-08 21:43:45 +00:00
fischman@webrtc.org
24c1778651 Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." 
Makes stopping flakier for some reason :/

BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:24:40 +00:00
jiayl@webrtc.org
b8f582591f Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. 
BUG=crbug/374457
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:42:00 +00:00
fischman@webrtc.org
171d94177b AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. 
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:22:37 +00:00
fischman@webrtc.org
b464618c84 Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 20:13:49 +00:00
fischman@webrtc.org
9512719569 AppRTCDemo(android): support app (UI) & capture rotation. 
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.

BUG=2432
R=glaznev@webrtc.org, henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:40:44 +00:00
fischman@webrtc.org
42694c5937 VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs. 
Since VCI::IF() fires a callback it risks a call back into VCI on the same
stack.  Failing to acquire _apiCs before _callbackCs means this is a lock
inversion and deadlock results.  By acquiring _apiCs first no lock inversion
occurs and the deadlock is removed.

BUG=3434
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:28:28 +00:00
wu@webrtc.org
94454b71ad Fix the chain that propagates the audio frame's rtp and ntp timestamp including: 
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:34:08 +00:00
tina.legrand@webrtc.org
65d61c3924 Opus send rate overflows if over 65 kbps 
The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 13:42:51 +00:00
bjornv@webrtc.org
b51d3ea593 Revert 6341 "Fixes and enables SystemDelayTests." 
> Fixes and enables SystemDelayTests.
> 
> The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
> This CL checks if it is in use.
> 
> BUG=3445
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12689005

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 13:41:33 +00:00
bjornv@webrtc.org
1f971b5788 Fixes and enables SystemDelayTests. 
The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
This CL checks if it is in use.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 10:58:55 +00:00
henrik.lundin@webrtc.org
2f816bbae7 NetEq: Add thread annotation to const scoped_ptrs 
Since the objects pointed to are not const, only the pointer to them,
they too must be accessed under lock.

Move the crit_sect to above the variables it is protecting.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12679006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 10:37:13 +00:00
bjornv@webrtc.org
aafd7a88c5 The correct fix of workaround in r6261. 
The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c

BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org, glaznev@webrtc.org, kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:53:51 +00:00
bjornv@webrtc.org
edbe886a0b common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND 
This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.

BUG=3348,3353
TESTED=trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:42:53 +00:00
stefan@webrtc.org
ef92755780 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:25:29 +00:00
henrike@webrtc.org
e6e139159f Android: cleanup gtest_target_type conditions. 
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 20:46:50 +00:00
solenberg@webrtc.org
c6db88b0cf Make it possible to build webrtc for arm64. 
- Bump revision of protobuf lib
- Remove -Wextra for arm64 gcc targets (warnings in stlport)
- Add MemoryBarrier implementation in single_rw_fifo.cc.
- [pending 15619004]: Bump revision of /deps/tools/android to get md5sum_bin for arm64.

BUG=chromium:354405,chromium:354539
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 17:15:42 +00:00
bjornv@webrtc.org
147f4fe3c0 Disables SystemDelayTest.CorrectDelayDuringDrift on Android 
Should have been part of https://webrtc-codereview.appspot.com/19629004/

BUG=3445
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 13:17:58 +00:00
bjornv@webrtc.org
b616e1211f Disables some modules_unittests on Android. 
BUG=3445
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 12:12:58 +00:00
andresp@webrtc.org
4436b4436a Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 09:05:30 +00:00
mflodman@webrtc.org
19fc09efba Adding missing break in media_file_utility.cc. 
There has been no reports of problems, but adding this to get it correct.

Review URL: https://webrtc-codereview.appspot.com/19599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 05:21:56 +00:00
marpan@webrtc.org
4ef254f781 Enable videoprocessor_integrationtest tests on android. 
R=kjellander@webrtc.org, stefan@webrtc.org
TBR=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/15599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 16:42:03 +00:00
turaj@webrtc.org
ddc6bc9347 Revert 6312 "Re-enable AudioCodingModuleMtTest" 
An example of botbreakage is http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/1807

> Re-enable AudioCodingModuleMtTest
> 
> Increase timeout and decrease test length. Also fixing a bug in the
> test, and make sure the test aborts if fatal failure occurrs.
> 
> BUG=3426
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13579005

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 15:25:34 +00:00
henrik.lundin@webrtc.org
8d13cd1956 Re-enable AudioCodingModuleMtTest 
Increase timeout and decrease test length. Also fixing a bug in the
test, and make sure the test aborts if fatal failure occurrs.

BUG=3426
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 12:53:21 +00:00
kwiberg@webrtc.org
8e4401b5a0 Reformat integer accessors to look like their float counterparts 
The new format is at least as easy to read, and takes less space.

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 10:04:13 +00:00
kwiberg@webrtc.org
c0035a67a1 Remove an optimization that's no longer worth the extra complexity it causes 
The data_ optimization was a way to operate on the data directly
instead of copying it, applicable in the mono, non-float case. Since a
few audio_processing steps are already using floats (with more
hopefully to come), we don't end up benefiting from the optimization
anyway, so we might as well remove it.

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:10:06 +00:00
solenberg@webrtc.org
a28c697d93 - Get rid of 'using' from .h 
- Add parenthesis to make order of evaluation clearer.

BUG=
R=minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:22:33 +00:00
henrik.lundin@webrtc.org
2bd032e11c Disable MouseCursorMonitorTest 
Last attempt reverted. Trying again in a different way.

This CL effectively reverts r6300.

BUG=3245
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/20549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 14:52:34 +00:00
henrik.lundin@webrtc.org
4ecae6e753 Disable MouseCursorMonitorTest.FromScreen 
The test is flaky.

BUG=3245
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 14:17:06 +00:00
henrik.lundin@webrtc.org
fe41a8f68d Adding thread annotations to parts of Audio Coding Module 
Picking some low-hanging fruit. Add annotations for acm_crit_sect_ that
do not require lock changes. Also adding annotations for callbacks.

BUG=3401
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 11:45:26 +00:00
bjornv@webrtc.org
2812b59acd Re-enables CommonFormats test for Android. 
It seems like this was a one time only and not a flaky test.

BUG=3376
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 11:27:29 +00:00
fischman@webrtc.org
360507b12b VideoCaptureAndroid: don't synchronized on camera thread. 
BUG=3421
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:17:38 +00:00
andrew@webrtc.org
1fddd6185d Add a Reset() method to AudioFrame. 
This method is introduced to try to avoid inconsistent resetting of
AudioFrame members to default/uninitialized values.

Use it at the call points of DownConvertToCodecFormat(). Results in the
following minor functional changes:
- speech_activity_ is set to its uninitialized value. AFAICT, this
member isn't used at all in the capture path.
- timestamp_ is switched from -1 to 0. This member doesn't appear to be
used either in the capture path, but left a TODO for wu to change the
default value to better represent the uninitialized state.

Bonus: Don't copy the frame on error in RemixAndResample(). An error
indicates a logical fault (as pointed out by the asserts) that we should
not attempt to recover from.
BUG=3111
R=turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 17:28:50 +00:00
andrew@webrtc.org
af48aaadf4 Disable AudioCodingModuleMtTest due to memcheck and tsan failures. 
This is a new test; the failures are not due to a change in underlying code.

TBR=henrik.lundin
BUG=3426

Review URL: https://webrtc-codereview.appspot.com/19589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 17:11:15 +00:00
henrik.lundin@webrtc.org
288bd15db8 Multi-threaded test for Audio Coding Module 
This CL adds a basic multi-threaded extention of the ACM unit test.
The test has three threads. One thread adds raw audio to the sender
side and encodes it. The next thread adds encoded RTP packets to the
receiver. The last thread pulls decoded audio out of the receiver.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 13:00:35 +00:00