7dccce3948
Revert 6048 "Implement the Windows screen capturer using the Mag..."
...
> Implement the Windows screen capturer using the Magnification API.
> The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
>
> BUG=2789
> TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
> R=sergeyu@chromium.org , wez@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/12149004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6052 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 11:17:26 +00:00
b235c56017
Implement the Windows screen capturer using the Magnification API.
...
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org , wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6048 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-03 00:16:29 +00:00
e44a84d851
Only clamp to 16 kHz when AECM is enabled.
...
Otherwise we could needlessly downsample to 16 kHz (rather than 32 kHz)
when HW AEC is used.
BUG=3259
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6033 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 18:58:23 +00:00
65f933899b
Fix constness of AudioBuffer accessors.
...
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.
Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 16:44:13 +00:00
9bd49becc1
Fix a data race in ACM1 when audio is pulled.
...
BUG=chromium:348511
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6026 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 20:27:45 +00:00
f2aafe4355
Added include of assert.h for files calling assert but missing the include.
...
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:54:17 +00:00
ceffdbc371
Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()
...
R=henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:11:07 +00:00
0300939484
Disable failing GoogleWifiTrace3Mbps.
...
Disables BweFeedbackTest.GoogleWifiTrace3Mbps instead of
BweSimulation.GoogleWifiTrace3Mbps.
TBR=stefan@webrtc.org
BUG=3277
Review URL: https://webrtc-codereview.appspot.com/20389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6017 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 15:25:59 +00:00
9353e6bc55
Disable GoogleWifiTrace3Mbps.
...
Breaks bots, according to stefan@ there's a missing file for this test
to run.
TBR=stefan@webrtc.org
BUG=3277
Review URL: https://webrtc-codereview.appspot.com/13409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6016 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:49:56 +00:00
dfe2a1c995
Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6015 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:21:42 +00:00
97e67cb476
Fix iOS assembly compile error.
...
In the roll of
https://webrtc-codereview.appspot.com/13369007
the fix in transform_neon.S was incorrectly removed
assuming it was only affecting Android when rolling to
265795. This CL fixes the iOS build when rolled to
266514.
Error looks like:
[893/2157] CC obj/webrtc/modules/audio_coding/codecs/isac/main/source/iSAC.entropy_coding.o
FAILED: /Volumes/data/b/build/goma/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang -MMD -MF obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o.d -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DDISABLE_NACL -DCHROMIUM_BUILD -DUSE_LIBJPEG_TURBO=1 -DENABLE_CONFIGURATION_POLICY -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DENABLE_EGLIMAGE=1 -DCLD_VERSION=1 -DENABLE_SPELLCHECK=1 -DDISABLE_FTP_SUPPORT=1 -DWEBRTC_RESTRICT_LOGGING -DWEBRTC_MODULE_UTILITY_VIDEO -DWEBRTC_ARCH_ARM -DWEBRTC_ARCH_ARM_V7 -DWEBRTC_ARCH_ARM_NEON -DWEBRTC_POSIX -DWEBRTC_MAC -DWEBRTC_IOS -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DNDEBUG -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DNS_BLOCK_ASSERTIONS=1 -D_FORTIFY_SOURCE=2 -I../.. -I../.. -I../../webrtc -I../../webrtc/common_audio/resampler/include -I../../webrtc/common_audio/signal_processing/include -I../../webrtc/common_audio/vad/include -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk -Os -gdwarf-2 -fvisibility=hidden -Werror -Wnewline-eof -miphoneos-version-min=6.0 -arch armv7 -Wall -Wendif-labels -Wextra -Wno-unused-parameter -Wno-missing-field-initializers -Wheader-hygiene -Wno-c++11-narrowing -Wno-char-subscripts -Wno-unneeded-internal-declaration -Wno-covered-switch-default -Wstring-conversion -Wno-deprecated-register -Wno-absolute-value -Wno-selector-type-mismatch -std=c99 -fcolor-diagnostics -c ../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S -o obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:45:11: error: immediate expression for mov requires :lower16: or :upper16
mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
^
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:458:11: error: immediate expression for mov requires :lower16: or :upper16
mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
in
http://build.chromium.org/p/client.webrtc/builders/iOS%20Release/builds/911/steps/compile/logs/stdio
TBR=ajm
TEST=ios trybots passing tryjob based on r6010.
BUG=
Review URL: https://webrtc-codereview.appspot.com/12439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 10:25:30 +00:00
59343ee3d8
Roll chromium_revision 260462:266514
...
Unfortunately needs to introduce yet another workaround
script for the Visual Studio toolchain download.
This will resolve the failures with our Dr Memory Full bot
(see https://code.google.com/p/chromium/issues/detail?id=366637#c2
for details). Long term, I'm considering a better approach
than using the added gclient solution pointing at
svn://svn-mirror.golo.chromium.org/chrome/trunk/deps/third_party/drmemory/drmemory.DEPS
i.e. add an entry that we roll separately in our DEPS file
instead. However, the Dr Memory team assured that changes
in their reporting format like this are rare.
Thanks fischman@ for the video_render.gypi fix!
Thanks kma@ for the transform_neon.S fix even if it turned out
not to be needed right now (probably will come back).
BUG=chromium:366637
TEST=git try -t compile
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13369007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6010 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 09:36:40 +00:00
acf15dc90f
Remove Version method from ACM1
...
BUG=2996
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 09:25:21 +00:00
70e53fa34d
Remove ACM1 and NetEq3 related targets from modules.gyp
...
Make necessary changes to compile.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6008 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:58:46 +00:00
fdf2053787
Remove AudioCodingModuleFactory
...
These were no longer used anywhere in the code.
BUG=2996
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6007 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:22:14 +00:00
0bc9b5a5a7
Add clock to ACM config struct
...
The purpose is to clean up the ACM interface a bit. This is a
follow-up of a comment in http://review.webrtc.org/13379004/ .
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6006 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:09:31 +00:00
059488f2ea
AEC: Startup phase only runs if reported_delay_enabled
...
TESTED=trybots, modules_unittests
R=aluebs@webrtc.org , andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20379005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6005 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:09:26 +00:00
82a045aae0
APM: limit native sample rate to 16kHz on mobile.
...
Required by AECM which assert-fails on higher sample rates.
BUG=3259
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6002 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 17:26:32 +00:00
497ff21fad
Using realpath instead of android_src in Android webview
...
BUG=367235
R=andrew@webrtc.org , torne@chromium.org
Review URL: https://webrtc-codereview.appspot.com/20369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6000 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 15:46:46 +00:00
494aa0e93d
AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h
...
These enums are noly used internally in aec_core.c and it makes more sense to put them in aec_core_internal.h
TESTED=trybots
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19379005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5995 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:42:27 +00:00
8dfe8ff590
Disable capture test for FrameRate on Windows.
...
Flaky on Windows, has been for a while.
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=3270
Review URL: https://webrtc-codereview.appspot.com/19389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5994 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:27:36 +00:00
e772c71743
Introduce a config struct for AudioCoding module
...
The config struct currently contains the module ID, and the NetEq
config struct, but will be extended in the future. The purpose of this
change is to expose certain NetEq settings to the ACM interface.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5993 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 10:16:57 +00:00
12a34247a4
Fix the NetEq build
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5990 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:36:35 +00:00
116ed1d4f0
Include buffer size limits in NetEq config struct
...
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.
The old constants governing the packet buffer limits are deleted.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:20:04 +00:00
b08bbf57a6
Add henrik.lundin as owner in AudioCoding module
...
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5988 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:15:35 +00:00
8f69330310
Replace scoped_array<T> with scoped_ptr<T[]>.
...
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar ...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
0175d76c72
Fix leak in remote bitrate estimator tests introduced in r5980
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5981 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 11:38:57 +00:00
4f616a02bd
Support for simulating multiple independent flows in a network.
...
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 10:59:24 +00:00
cc1ba15fe7
Returns a NULL frame on all platforms if the captured window is closed.
...
Part of the fix for crbug/360181.
On Mac/Linux, it previously continues capturing even if the window is closed.
Now it stops by returning a NULL frame.
On Windows, it used to stop capturing when the window is minimized. Now fixed to match other platforms.
Note: the crbug still needs a chrome side fix to close the notification bar.
This fix only stops the stream (i.e. stream onended event fired).
BUG=crbug/360181
TESTED=manually tested in Chrome
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12329007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5977 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 23:45:56 +00:00
cd70119a10
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
...
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 22:10:24 +00:00
93fd25c20c
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
...
* Cast rtp header extension to int in log in rtp_utility.cc.
BUG=3237
TEST=try bots
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 20:33:08 +00:00
439a4c49f9
Add an output capacity parameter to ACMResampler::Resample10Msec()
...
Also adding a unit tests to make sure that a desired output frequency
of 0 passed to AudioCodingModule::PlayoutData10Ms() is invalid.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 19:05:33 +00:00
103657b484
Add keyboard channel support to AudioBuffer.
...
Also use local aliases for AudioBuffers for brevity.
BUG=2894
R=aluebs@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 18:28:56 +00:00
d57b8149c2
Fix the Android compilation (better structure for NetEq test libs)
...
This change should make the Android targets compile again. The reason
for the failure was a highly dubious structure in the gypi files. With
this fix, the structure is somewhat cleaner. Still room for improvement.
BUG=3254
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5972 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 13:19:04 +00:00
0a2277448e
Fixing a bug in ACM2 where the output frame energy was incorrectly set
...
The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).
Implemented a unit test to verify ACM output frame.
BUG=3255
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 08:11:39 +00:00
f26c9e8369
Use unique filenames in AudioProcessingTests for parallelization.
...
TBR=bjornv
TESTED="gtest-parallel -w 32 --gtest_filter=*AudioProcessingTests*
out/Debug/modules_unittests" passes.
Review URL: https://webrtc-codereview.appspot.com/14369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5968 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 03:46:46 +00:00
e9d3760d5c
AEC: Adds a reported_delay_enabled_ flag
...
Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.
TESTED=trybots, modules_unittest
R=aluebs@webrtc.org , andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12349005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 13:20:07 +00:00
46b31b17df
Restore sample_rate_hz() until Chromium is updated to not use it.
...
TBR=bjornv
TESTED=Chromium builds against webrtc head.
BUG=2894
Review URL: https://webrtc-codereview.appspot.com/12349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5962 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 03:33:54 +00:00
ddbb8a2c24
Support arbitrary input/output rates and downmixing in AudioProcessing.
...
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org , bjornv@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 21:00:04 +00:00
d59359af4d
Remove 44.1 kHz workaround from the iOS AudioDevice.
...
Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.
This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.
BUG=1395
R=fischman@webrtc.org , henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 18:07:49 +00:00
20c71fd1dc
Fix a bug in AcmReceiver::NetworkStatistics
...
One of the variables were not copied between the structs.
BUG=2996
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5956 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 10:11:21 +00:00
0c1444c748
Create ACM2 instance when calling AudioCodingModule::Create
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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
5964fe0f86
audio_processing: DestroyHandle() now returns void
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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
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* 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.
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- 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.
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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."
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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
be7585b150
Fix multi-monitor support in the screen capturer for Mac.
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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.
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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
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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
