e5251ad63c
Integrate send-side BWE into simulation framework.
...
BUG=4173
R=mflodman@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8123 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 10:10:53 +00:00
cfd82dfc11
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
...
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 09:39:59 +00:00
3dd33a6787
Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes.
...
BUG=crbug:425925
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8121 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 09:12:23 +00:00
fbd37bd737
Make iSAC SWB own its decoder
...
A bug in the ACM codec database caused iSAC-swb to behave differently
from iSAC-wb and -fb. With this fix, all iSAC codecs behave the same
with respect to decoder ownership.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8120 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 08:16:29 +00:00
e65d9d974c
Fix an unitialized variable warning.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35819004
Patch from Sebastien Marchand <sebmarchand@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8118 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 22:05:12 +00:00
c429b824b3
GN: Prepare to remove webrtc_base target
...
Keep the webrtc_base target temporarily while waiting for
Chromium to pick up this revision. Then we'll update Chromium
and remove the webrtc_base target for real.
This should have been a part of https://code.google.com/p/webrtc/source/detail?r=7140
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8117 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 20:22:33 +00:00
c78d81ae89
Re-land "Support 48kHz in AEC"
...
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
Original: https://webrtc-codereview.appspot.com/28319004/
Reverted: https://webrtc-codereview.appspot.com/33949004/
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8116 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 19:10:55 +00:00
e81c5d6d7e
Fix TransientDetectorTest in modules_unittests on Android ARM64
...
BUG=webrtc:4200
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8115 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 18:01:28 +00:00
11af039590
Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.
...
BUG=4199
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8114 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 14:22:39 +00:00
df7b65ba01
Change CreateOrGetReportBlockInformation to have one return path.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 13:07:04 +00:00
9ffd8fe96b
Indentation changes.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 08:22:50 +00:00
7dba7860c7
Setting Opus target application.
...
This CL is to allow to set Opus target application at the creation of an encoder.
According to Opus spec, there are three applications:
OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY
BUG=
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 16:01:50 +00:00
853049fa30
Move internal capture+render to build_with_chromium==0 condition
...
This will avoid errors related to DirectX not being found
for Chromium bots (mainly GN, but it's safest to do the same
changes for GYP since they also make sense there as GYP generation
go slightly faster without having to process those targets).
Thanks to vchigrin@yandex-team.ru for originally suggesting
this being fixed in
https://webrtc-codereview.appspot.com/37639004/
TESTED=
Successfully ran:
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dbuild_with_chromium=1
and trybots.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8102 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 11:40:45 +00:00
ee0c100d54
Revert 8080 "Support 48kHz in AEC"
...
> Support 48kHz in AEC
>
> Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
> Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
>
> BUG=webrtc:3146
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28319004
TBR=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8100 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 10:22:49 +00:00
9691b36995
Cleanup for Rtp Rtcp API test.
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 05:42:52 +00:00
474e36e623
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
...
The previous CL was reverted for two reasons:
- Added a static initializer because std::string.
- Landed before the corresponding chromium CL, which has now been landed.
BUG=crbug:425925
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8094 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 15:44:47 +00:00
a32d15448d
Disable tests failing on Android ARM64 (Nexus9).
...
BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.
Review URL: https://webrtc-codereview.appspot.com/33919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 12:46:01 +00:00
a1aea10af2
Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."
...
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8085 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-16 13:52:52 +00:00
4ba1e44ff0
Remove unnecessary remote bitrate estimator build rule which serves no purpose.
...
BUG=4185
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8084 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-16 07:50:17 +00:00
64d3c4b9ac
Support 48kHz in AEC
...
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8080 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 19:52:05 +00:00
d82f55d2a7
Only adapt AGC when the desired signal is present
...
Take the 50% quantile of the mask and compare it to certain threshold to determine if the desired signal is present. A hold is applied to avoid fast switching between states.
is_signal_present_ has been plotted and looks as expected. The AGC adaptation sounds promising, specially for the cases when the speaker fades in and out from the beam direction.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28329005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8078 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 18:07:21 +00:00
3e42a8a56a
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
...
BUG=crbug:425925
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8076 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 14:45:27 +00:00
1f67b53c88
Remove dual stream functionality in ACM
...
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 09:36:30 +00:00
0800db74b9
Add percentage of fec packets and recovered media packets to histogram stats:
...
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 07:40:20 +00:00
6c3855258d
Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
...
This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Review URL: https://webrtc-codereview.appspot.com/36689004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 02:56:06 +00:00
5a92b78e86
Add beamforming to audioproc_float utility.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8069 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 01:28:36 +00:00
6b6301588e
Move ring_buffer to common_audio.
...
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 00:09:53 +00:00
693e01c910
Fix searching for DirectX SDK during GN build.
...
Before that GN just checked for DXSDK_DIR environment variable.
GYP does more and checks registry, let's do the same in GN.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37599004
Patch from Vyacheslav Chigrin <vchigrin@yandex-team.ru >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8066 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 21:25:25 +00:00
387841ac5c
Improved fairness simulation by starting the flows 20 seconds apart.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8062 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:45:29 +00:00
f18fba2f7b
Implement SimulcastEncoderAdapter support.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/37589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:26:23 +00:00
2ebfac5649
Remove COMPILE_ASSERT and use static_assert everywhere
...
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org , andrew@webrtc.org , hellner@chromium.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
d6e84d9d13
Always copy processed audio to output buffer in ProcessStream.
...
In the old AudioFrame ProcessStream API, input and output buffers were shared.
Now that the buffers are distinct, the input must be copied to the
output even when no processing occurred.
R=andrew@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=78de5010d167d1e375e05d26177aad43c2e2de08
Review URL: https://webrtc-codereview.appspot.com/41459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8052 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 01:33:54 +00:00
0b0c24177b
Only return Rtx mode in RTXSendStatus().
...
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38569004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 14:15:15 +00:00
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
46323b3786
Remove useless AudioProcessing::Create() overload.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8046 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 06:48:06 +00:00
16825b1a82
Use int64_t more consistently for times, in particular for RTT values.
...
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , holmer@google.com , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
a7add19cf4
audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348,3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8044 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:12:29 +00:00
2a26734f04
Partial revert of r7396
...
This change reverts a small part of what was done in r7396. It seems
like that change uncovered another issue with NEON.
BUG=4177,chrome-os-partner:31534
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8043 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 20:52:21 +00:00
a525c98ca5
Fix parallelizability in ApmTests.
...
Using temporary filenames instead of fixed ones permits them to run in
parallel.
BUG=chromium:445880
R=andrew@webrtc.org , kjellander@webrtc.org
TEST=third_party/gtest-parallel/gtest-parallel -r100 -w100 out-asan/out/Debug/modules_unittests --gtest_filter=*ApmTest*:*CommonFormats*
Review URL: https://webrtc-codereview.appspot.com/35709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8041 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 17:31:18 +00:00
c14e3572c6
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
...
BUG=3353, 1133
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8037 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:50:52 +00:00
2693a54614
Add WEBRTC_BEAMFORMER define to BUILD.gn
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8034 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 23:26:13 +00:00
8f27fcce79
Revert 8028 "Support associated payload type when registering Rt..."
...
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org , stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com >.
TBR=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 20:22:46 +00:00
f3fd8e7cdf
Add NEON intrinsics version for transform_neon
...
WebRtcIsacfix_Time2SpecNeon and WebRtcIsacfix_Spec2TimeNeon are added.
TransformTest in modules_unittests is passed on ARM32/ARM64 platform.
Initially reviewed here:
https://webrtc-codereview.appspot.com/36449004/
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I0920ff66a0a0f529707fd7e6619f91e271a47019
Review URL: https://webrtc-codereview.appspot.com/31309004
Patch from Yang Zhang <yang.zhang@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8030 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 18:29:37 +00:00
2a169640a3
Support associated payload type when registering Rtx payload type.
...
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.
BUG=4024
R=pbos@webrtc.org , stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26259004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 15:16:10 +00:00
8649fed1b8
GN: Fix Windows build.
...
This required a tiny include fix in
src/third_party/winsdk_samples/src
which was committed in
https://code.google.com/p/webrtc/source/detail?r=7951
This incorporates contribution from vchigrin@yandex-team.ru
in https://webrtc-codereview.appspot.com/29299004/
BUG=261,1348,4105
R=pbos@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 21:22:01 +00:00
758d6d431e
audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8025 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 17:52:56 +00:00
dec649cbab
audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with *
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8024 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 17:34:33 +00:00
5e5b32706a
audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8023 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 17:25:34 +00:00
3663fb08ff
Reenable dlclose() for InternalUnloadDll on TSan.
...
Upstream TSan bug has been fixed and dlclose() no longer needs to be
excluded.
R=henrika@webrtc.org
BUG=3895
Review URL: https://webrtc-codereview.appspot.com/30099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8016 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 18:02:39 +00:00
