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
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
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
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
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
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
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
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
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
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
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
84d84471f5
Minor fixes regarding accumulator usage on MIPS platforms.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33729004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7979 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 17:08:44 +00:00
1090a6eccf
Remove obsolete target_arch == armv7.
...
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.
BUG=3906
R=kjellander@webrtc.org , tkchin@webrtc.org , zhongwei.yao@arm.com
Review URL: https://webrtc-codereview.appspot.com/38379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 21:36:18 +00:00
eed7a22bbf
Make an AudioEncoder subclass for iSAC redundant encoding
...
Adding unit test, too.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:52:36 +00:00
cab1291745
Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder
...
Re-enable the test and explicitly call delete on red, even though the
test should die in the AudioEncoderCopyRed cunstructor. Apparently,
things work a little differently under memcheck.
BUG=4108, 3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 06:58:42 +00:00
eb544460e4
Rename _t struct types in audio_coding.
...
_t names are reserved in POSIX.
R=henrik.lundin@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/34509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 15:23:29 +00:00
e728ee03ba
Remove or rename typedefs with _t prefixes.
...
_t prefixes are reserved for additional typenames in POSIX.
R=henrik.lundin@webrtc.org , hta@webrtc.org , stefan@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/36559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 13:43:55 +00:00
e102e8147b
Enable the iSACfix AudioDecoder test (and make it work again)
...
As far as I can tell, the test should have been enabled again once
https://code.google.com/p/webrtc/issues/detail?id=1353 was fixed, but
it wasn't, and has rotted a bit as a result. I'm not sure why the
number of encoded bytes have changed, but the output seems to be
correct (EncodeDecodeTest encodes, decodes, and compares the result
with the original).
The DecodePlc change is necessary because r7912 added support for that
to the iSACfix AudioDecoder.
BUG=1353, 3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 07:30:23 +00:00
a32487f97b
Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder
...
Fails linux memcheck.
BUG=4108
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:04:55 +00:00
c1c9291e9b
Make an AudioEncoder subclass for RED
...
This class only supports the simple case of payload duplication. That
is, one single encoder is used, and the redundant payload is a one-step
delayed payload.
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
88bdec8c3a
AudioEncoder subclass for iSACfix
...
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:49:37 +00:00
0b1534c52e
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
...
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
b413a30097
Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
...
This intrinsics version gives bit-exact result as the current assembly
neon code. And the performance is 38% better than current assembly
neon version, 5.92 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I257e33ef6d634a519fd71adc4f52b06dd655bd9d
Review URL: https://webrtc-codereview.appspot.com/32749004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:23:49 +00:00
3b79daff14
Moving encoded_bytes into EncodedInfo
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 13:31:24 +00:00
0ca768b131
Adding DTX to WebRTC Opus wrapper (relanding).
...
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.
See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/
Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 16:09:35 +00:00
817e50dd7d
Make an AudioEncoder subclass for PCM16B
...
The implementation depends on AudioEncoderPcm to reduce code
duplication.
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
b3ad8cf6ca
Make an AudioEncoder subclass for iSAC
...
BUG=3926
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
abe3f1879c
Checking whether ACM uses codec internal or WebRTC DTX.
...
It was not clear how one could know if ACM is using DTX from WebRTC or codec internal DTX.
This CL makes better use of IsInternalDTXReplacedWithWebRtc() which was designed for G.729 to export such information.
Before
IsInternalDTXReplacedWithWebRtc() gives true only if codec == G729 and G729's internal DTX is replaced with WebRTC DTX.
Now
IsInternalDTXReplacedWithWebRtc() gives true also when codec does not have internal DTX, i.e., must use WebRTC DTX, which is much more logical.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:53:21 +00:00
55d42c32a4
DCHECK: Reference condition parameter in release builds
...
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as
int x = ...
DCHECK_EQ(x, 17);
R=andrew@webrtc.org
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7858
and reverted: https://code.google.com/p/webrtc/source/detail?r=7859
Review URL: https://webrtc-codereview.appspot.com/31169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:32:30 +00:00
d8ca723de7
Remove CELT support from audio_coding.
...
R=henrik.lundin@webrtc.org , juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 11:49:13 +00:00
3cd26b677a
Revert r7858 ("DCHECK: Reference condition parameter in release builds")
...
Apparently Visual Studio is cleverer than I am at figuring out what
local variables are actually unused.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7859 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:57:14 +00:00
3148060e61
DCHECK: Reference condition parameter in release builds
...
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as
int x = ...
DCHECK_EQ(x, 17);
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7858 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:45:47 +00:00
ff1a3e36bd
Make an AudioEncoder subclass for comfort noise
...
BUG=3926
R=bjornv@webrtc.org , kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 07:29:08 +00:00
19dd129c69
Revert 7846 "Adding DTX to WebRTC Opus wrapper"
...
> Adding DTX to WebRTC Opus wrapper
>
> This is a step toward adding Opus DTX support in WebRTC.
>
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
>
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
>
> We transmit the first 1-byte packet to let decoder be in-sync
>
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13219004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 15:11:15 +00:00
4321f175f1
Adding DTX to WebRTC Opus wrapper
...
This is a step toward adding Opus DTX support in WebRTC.
Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
We transmit the first 1-byte packet to let decoder be in-sync
BUG=webrtc:1014
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 13:27:39 +00:00
1784d7cfad
Adding an codec interal CNG test in NetEq.
...
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7843 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:46:39 +00:00
e04a93bcf5
Move the AudioDecoder interface out of NetEq
...
It belongs with the codecs, next to the AudioEncoder interface.
R=andrew@webrtc.org , henrik.lundin@webrtc.org , kjellander@webrtc.org
Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
8911bc52f1
Add AudioEncoder::Max10MsFramesInAPacket
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:15:55 +00:00
130fef89dd
Bugfix in AudioDecoderTest
...
When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:07:59 +00:00
fcbe36a1d9
Add const qualifier to WebRtcPcm16b_Encode
...
BUG=909
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 18:26:49 +00:00
a1ef7bfa15
ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable.
...
Ideally, this is a stopgap fix until ATTRIBUTE_UNUSED can be given a
proper definition.
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7830 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:53:10 +00:00
cb858ba397
Make an AudioEncoder subclass for iLBC
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@google.com
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:11:44 +00:00
33ccdfa1f5
Relanding r7807.
...
r7807 was reverted to be excluded from the cause of a failure.
It has been verified and can reland now.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 12:14:12 +00:00
52bc4f4797
Revert 7807 "Removing unused opus wrapper APIs."
...
> Removing unused opus wrapper APIs.
>
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
>
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
>
> BUG=
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28139004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 11:00:50 +00:00
e54a6342dd
Removing unused opus wrapper APIs.
...
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
WebRtcOpus_DecodePlcMaster/Slave() are also removed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 08:47:25 +00:00
3a52458237
add WebRtcIsacfix_AutocorrNeon's intrinsics version
...
The modification only uses the unique part of the
WebRtcIsacfix_AutocorrC function. Pass FiltersTest.AutocorrFixTest test
on both ARMv7 and ARM64, and the single function performance is similar
with original assembly version on different platforms. If not
specified, the code is compiled by GCC 4.6. The result is the "X
version / C version" ratio, and the less is better.
| run 100k times | cortex-a7 | cortex-a15 |
| use C as the base on each | (1.2Ghz) | (1.7Ghz) |
| CPU target | | |
|----------------------------+-----------+------------|
| Neon asm | 24% | 23% |
| Neon intrinsics (GCC 4.6) | 33% | 32% |
| Neon intrinsics (GCC 4.8) | 27% | 27% |
BUG=3850
R=andrew@webrtc.org , jridges@masque.com
Change-Id: Id6cd0671502fadbebd10b1f5493f5b16c988286f
Review URL: https://webrtc-codereview.appspot.com/27999004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7802 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 21:58:18 +00:00
8dc21dc238
Rename internal AudioEncoder::Encode method to EncodeInternal
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 20:36:03 +00:00
3800e13a3a
Revert r7798 ("Move the AudioDecoder interface out of NetEq")
...
Apparently, it caused all sorts of problems I don't have time to
straighten out right now.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 16:28:17 +00:00
00ba1a7dfd
Move the AudioDecoder interface out of NetEq
...
It belongs with the codecs, next to the AudioEncoder interface.
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 14:23:23 +00:00
