d144bb6812
Let A/V sync test use default AudioCoding module
...
This test used to run with both ACM1 and ACM2, to verify sync with both
versions of the module. ACM1 (and NetEq3) is now being deprecated,
wherefore this test should now use the default one (i.e., ACM2).
BUG=2996
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5953 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:36:33 +00:00
0c1444c748
Create ACM2 instance when calling AudioCodingModule::Create
...
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5952 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:18:42 +00:00
372ae83228
Reland "Make VoiceEngine choose ACM2 by default""
...
This cl was originally committed as r5923, but was reverted in r5926
due to a blocking bug (issue 3143). The blocking bug was resolved in
r5936.
BUG=2996
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 07:21:03 +00:00
5964fe0f86
audio_processing: DestroyHandle() now returns void
...
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.
BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org , aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 06:52:28 +00:00
2a796720f8
common_audio: VADFree() now returns void
...
* Files in audio_coding are not affected since they never use the return value.
* voice_detection in audio_processing does.
* Updated vad_unittest.cc
BUG=441
TESTED=trybots
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5948 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 04:45:35 +00:00
f5a33f145b
Resampler modifications in preparation for arbitrary audioproc rates.
...
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.
This is a prerequisite of:
http://review.webrtc.org/9919004/
BUG=2894
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:32:07 +00:00
3d9ec1fed4
Fix multi-monitor support in the screen capturer for Mac.
...
This feature was broken in r5471.
BUG=361919
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=5937
Review URL: https://webrtc-codereview.appspot.com/12109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:25:35 +00:00
7d055a6e63
Revert r5937 "Fix multi-monitor support in the screen capturer for Mac."
...
This would break when rolled in chromium because some code in
chromium depends on the code I changed in that change.
TBR=jiayl@webrtc.org
BUG=361919
Review URL: https://webrtc-codereview.appspot.com/12199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5940 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 23:45:38 +00:00
0daa8be9d6
Add Chromium's ScopedVector.
...
Trivial changes from the original excepting scoped_vector_unittest.cc,
diff here: https://paste.googleplex.com/6664017300946944
This is a prerequisite for:
http://review.webrtc.org/9919004/
TBR=henrike@webrtc.org
BUG=2894
Review URL: https://webrtc-codereview.appspot.com/12179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 21:20:54 +00:00
be7585b150
Fix multi-monitor support in the screen capturer for Mac.
...
This feature was broken in r5471.
BUG=361919
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5937 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 18:22:41 +00:00
a596a389ea
Fix iSAC/48000 issue with ACM2.
...
Registeration of iSAC into NetEq is through injecting and external AudioDecoder. This is because iSAC encoder and decoder need to share instances for bandwidth estimator to work. When external decoder is registerred, the sampling rate of the decoder had to be specified. iSAC/48000 decoder has a native sampling rate of 32000 Hz, but it has been registered as 48000 Hz decoder.
This CL fixing this issue by letting NetEq to obtain sampling rate of an external coder according to its existing database.
BUG=3143
TEST=voe_cmd_test,modules_unittest,try-bots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5936 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 23:30:49 +00:00
e57ae02327
WebRtcAecm_Process: Reduce code duplication
...
BUG=
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5930 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:28:33 +00:00
d2f366f28c
StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
...
The max value is ((2**15 - 1) + (2**15 - 1)) >> 1
== (2**16 - 2) >> 1
== 2**15 - 1
which doesn't overflow.
The min value is (-2**15 + -2**15) >> 1
== -2**16 >> 1
== -2**15
which doesn't overflow.
Since those two bracket all possible results, the call to
WebRtcSpl_SatW32ToW16 is redundant.
BUG=
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:17:39 +00:00
66803489f9
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=henrik.lundin@webrtc.org , juberti@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:45:01 +00:00
0f7375504a
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=juberti@webrtc.org , niklas.enbom@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:38:08 +00:00
e2e9abb3bc
Revert "Make VoiceEngine choose ACM2 by default"
...
The reason for reverting is that Issue 3143 should be resolved
first.
TBR=henrika@webrtc.org
BUG=3143
Review URL: https://webrtc-codereview.appspot.com/12119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5926 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:12:27 +00:00
adaf809612
Removing AudioCoding duplicate tests
...
Reverting to using one version of ACM in ACM tests.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:29:10 +00:00
6cec07f6a7
Make VoiceEngine choose ACM2 by default
...
The use of a factory for ACM will be removed in later CLs.
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:07:18 +00:00
c0a15b7ddc
Fix crashes due to dangling external decoder pointer.
...
When checking whether we need to release external decoder,
we have to do pointer comparison. We can't rely on payload
types, because payload types can be stale (e.g. before we
decode the first video frame after RegisterReceiveCodec).
This leaves a dangling pointer to external decoder, which
leads to crashes later, after we actually delete the
external decoder object.
This change has been verified in Chromecast code tree.
BUG=chromium:335539
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12049004
Patch from Sergey Volk <servolk@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 01:22:48 +00:00
c18729165a
Set include_internal_video_capture=1 for video_capture_tests
...
Having this override in the .gypi file avoids having to set it for the bots, which I think is best if we can avoid.
This CL also reverts r5869 so video_capture_tests are compiled for Android again.
BUG=2974,3152
TEST=Successfully ran:
git try -t compile
git try --bot=win_baremetal,mac_baremetal,linux_baremetal -t video_capture_tests
git try --bot=android_apk,android_apk_rel
Verified the change in webrtc/build/apk_tests.gyp makes the build compile successfully from a Chromium+WebRTC configured checkout for Android APK tests.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 12:59:49 +00:00
f927fd6481
Re-enable AGC tests:
...
* AgcConfigTest.HasCorrectDefaultConfiguration
* AgcConfigTest.DealsWithInvalidParameters
* AgcConfigTest.CanGetAndSetAgcStatus
* AgcConfigTest.HasCorrectDefaultRxConfiguration
* AgcConfigTest.DealsWithInvalidRxParameters
* AgcConfigTest.CanGetAndSetRxAgcStatus
* AudioProcessingTest.AgcIsOnByDefault
* AudioProcessingTest.CanEnableAgcWithAllModes
* AudioProcessingTest.RxAgcShouldBeOffByDefault
* AudioProcessingTest.CanTurnOnDigitalRxAcg
* AudioProcessingTest.CannotTurnOnAdaptiveAnalogRxAgc
BUG=webrtc:2784
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 11:58:18 +00:00
7de47bce12
Remove use of tmpnam.
...
This solves compilation with the Mac SDK 10.9.
BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org , henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
2c3f1abb69
Replace flooding logs in rtp_sender.cc with a comment.
...
Started occurring after:
https://webrtc-codereview.appspot.com/11129004
BUG=3153
R=andresp@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 21:26:34 +00:00
ca539bbed0
iOS: baby steps to being able to include_tests=1
...
- pull iossim in DEPS even when on mac (because bug 2152)
- fix audio_device_test_api.cc's use of bool instead of bool* (!)
- move unused-on-mobile message to non-mobile-only section of
hardware_before_streaming_test.cc
BUG=3185
R=kjellander@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 20:26:41 +00:00
7c6e3d188a
Moved voe_neteq_stats_unittest to audio_coding_module_unittest
...
The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.
BUG=2996
R=henrika@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:59:25 +00:00
6c75c98964
Propagate capture ntp timestamp from rtp to renderer.
...
Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.
TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:46:33 +00:00
2f8d5f3302
Check if a header extension is registered before updating it and fail silently if it's not.
...
BUG=
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 12:28:46 +00:00
6e105ede38
Make WebRTC Android examples build without sourcing envsetup.sh
...
The new recipes framework for configuring build explicitly sets the
GYP_DEFINES for Android builds instead of relying on the envsetup.sh script
which probably will be removed at some point in the future.
This causes our build to break since our Android examples relies on the
Android SDK being found using the ANDROID_SDK_ROOT environment variable.
A GYP variable 'android_sdk_root' exists and is set correctly by
common.gypi, which is what I'm using to pass this path correctly to these
tests.
The libjingle example is handled separately in
https://webrtc-codereview.appspot.com/11809004/
BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 08:35:00 +00:00
2c89b5cb27
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
35ead381f8
Adding a config struct to NetEq
...
With this change, the parameters sent to the NetEq::Create method are
collected in one NetEq::Config struct. The benefit is that it is easier
to set, change and override default values, and easier to expand with
more parameters in the future.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5902 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:49:17 +00:00
810acbc93e
New Packet and PacketSource classes for NetEq tests
...
These new classes are intended to replace the old NETEQTEST_RTPpacket
classes. The code in rtp_analyze.cc has been updated to use the new
classes; other test applications will follow.
BUG=2692
R=andrew@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:42:23 +00:00
5cf73962e6
Fix gyp for video_capture/ensure_initialized.cc.
...
This is a follow-up to
https://webrtc-codereview.appspot.com/11359004
which introduced an invalid dependency in the
chromium build when building without linker GC.
BUG=2974,3152,chromium:354405
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11789005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 17:26:31 +00:00
b9309beea4
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 14:12:50 +00:00
5692531f18
Added a new OnMoreData() interface which will not feed the playout data to APM.
...
BUG=3147
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 10:50:37 +00:00
8ce7c72456
Fix the captured screen rect conversion.
...
device_mode.dmPosition is already relative to the primary display's top-left, while the expected value of GetScreenRect() is also relative to the primary display's top-left.
TESTED=verified on Windows single monitor capturing and cursor capturing is fixed.
BUG=https://code.google.com/p/chromium/issues/detail?id=362631
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/11789006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 22:31:15 +00:00
8d1cdaa84e
NetEq changes.
...
BUG=
R=henrik.lundin@webrtc.org , minyue@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5889 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 18:47:55 +00:00
34c5da6b5e
Cleaned up logging in video_coding.
...
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.
BUG=3153
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 14:08:35 +00:00
8b2ec15d1e
Convert WEBRTC_TRACE to LOG in utility.
...
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 07:59:43 +00:00
22cf7472a0
Disable UsesTraceCallback
...
Ongoing removal of trace code is causing UsesTraceCallback to fail,
disabling it for now.
BUG=3157
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5882 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 14:39:22 +00:00
e6013bb7be
Fix loopback test for case where no constraint is given.
...
R=stefan@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5881 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 14:21:45 +00:00
2a770828d8
Remove usage of webrtc trace in video processing modules.
...
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 11:30:49 +00:00
0273fa98e0
Add ability to control peer connection constraints for the loopback test.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11419005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5879 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 09:40:16 +00:00
f93021430d
Remove self-assignment hacks that were added to avoid unused variable warnings.
...
Instead, appear to use the variables.
BUG=3152
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5877 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 21:19:55 +00:00
0569d93db7
Move a chatty creation log in neteq to LS_VERBOSE.
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R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5876 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 17:48:48 +00:00
f4357f3530
Make Android-APK compile in release again.
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BUG=3152
R=kjellander@webrtc.org
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5874 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 14:21:37 +00:00
8883a0f47f
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
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Landing https://webrtc-codereview.appspot.com/11419004/ manually.
TBR=niklase
BUG=none
Review URL: https://webrtc-codereview.appspot.com/11439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 13:04:12 +00:00
2e9d89cf77
Unbreak android APK buildbots by emptying the video_capture_tests_apk target.
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Needed until the bots start to specify include_internal_video_capture=1.
TBR=henrike@webrtc.org
BUG=3152
Review URL: https://webrtc-codereview.appspot.com/11479006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 02:34:50 +00:00
b0b135e4c2
VideoCaptureAndroid: support multiple frame-rates per resolution.
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Also enables running video_capture_tests_apk on the WebRTC/Chromium APK bots,
assuming GYP_DEFINES includes include_tests=1 and
include_internal_video_capture=1.
This required running VideoCaptureAndroid's camera capture on a dedicated thread, matching other platform's video_capture impls.
BUG=2974,3152
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5868 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 01:18:32 +00:00
74f6074ec1
Fix DesktopSize::is_empty() for the case when only width or only height is 0.
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BUG=crbug.com/358909
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/11479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 01:04:22 +00:00
a78a41f985
Move output_mixer_unittest.cc to utility_unittest.cc.
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This reflects a move of the tested code in:
https://webrtc-codereview.appspot.com/11019005/
TBR=xians
Review URL: https://webrtc-codereview.appspot.com/11449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-08 23:09:28 +00:00
