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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
420b2567f3
Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers.
...
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728 .
BUG=1811
R=asapersson@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 12:17:15 +00:00
a816180f93
Fixing a bug regarding VOE packet loss rate feedback to ACM
...
Phenomenon:
When packet loss rate was fed to a codec that does not implement packet loss adaptive encoding, VoE logs an error.
Reason:
The ACM function SetPacketLossRate(int rate) return -1 unnecessarily too often. It was intended for more severe errors like
1. codec is not ready
2. input rate is out of range
BUG=webrtc:3413
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 09:28:07 +00:00
6e732c6765
Revert 6272 "Update generated asm offsets scripts."
...
Revert since it fails webrtc-in-chromium Android bots.
> Update generated asm offsets scripts.
>
> Libvpx updated the unpack scripts to fix building dependencies.
>
> Roll libvpx 269083:273304
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> for the libvpx changes.
>
> BUG=377062
> R=andrew@webrtc.org , michaelbai@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/12579008
TBR=fgalligan@google.com
Review URL: https://webrtc-codereview.appspot.com/12649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6282 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 09:19:03 +00:00
21a5d449b7
Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21499006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6274 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 19:43:26 +00:00
2a8efa8971
Update generated asm offsets scripts.
...
Libvpx updated the unpack scripts to fix building dependencies.
Roll libvpx 269083:273304
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
for the libvpx changes.
BUG=377062
R=andrew@webrtc.org , michaelbai@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12579008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6272 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 17:08:34 +00:00
d6a0efdc86
VideoCaptureAndroid: quit & join the camera thread on stopCapture.
...
Also fix latent bug where setPreviewRotation() wouldn't hold
the lock while its delegate setPreviewRotationOnCameraThread()
was running, allowing the camera to be freed between the
null-check and the use.
BUG=3389
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6266 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 18:37:07 +00:00
f15c14be22
Echo canceler: Saturate output to guarantee it'll be in the allowed range
...
r6138 (https://webrtc-codereview.appspot.com/18399005/ ) somewhat
ill-advisedly removed the saturation step at the end of
aec_core.c:NonLinearProcessing(); this patch restores it.
BUG=
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6263 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 11:47:08 +00:00
0aa3ee661c
Better buffer size estimation in NetEq for redundant packets
...
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6260 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:48:01 +00:00
1b9df05c85
Revert 6257 "Rename neteq4 folder to neteq"
...
> Rename neteq4 folder to neteq
>
> BUG=2996
> R=turaj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12569005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:33:39 +00:00
637c55f45b
Add support of texture frames for video capturer.
...
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.
Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352
BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:00:51 +00:00
a90f6d67f7
Rename neteq4 folder to neteq
...
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 06:23:34 +00:00
27e884cf47
Disable MouseCursorMonitorTest due to flake on Windows.
...
TBR=sergeyu
BUG=3408
Review URL: https://webrtc-codereview.appspot.com/15589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6256 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 03:34:04 +00:00
89e8ffb395
Revert "Add support of texture frames for video capturer."
...
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.
Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.
BUG=chromium:362437
TBR=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 14:12:58 +00:00
efe15355ee
Add support of texture frames for video capturer.
...
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.
Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352
BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org , perkj@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 12:40:27 +00:00
