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186 Commits

Author SHA1 Message Date
andrew@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
andrew@webrtc.org
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
henrik.lundin@webrtc.org
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
minyue@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
henrik.lundin@webrtc.org
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
minyue@webrtc.org
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
minyue@webrtc.org
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
kwiberg@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
minyue@webrtc.org
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
minyue@webrtc.org
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
minyue@webrtc.org
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
andrew@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
henrik.lundin@webrtc.org
7f1dfa5b61 Adding a payload type to AudioEncoder objects 
The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 12:08:39 +00:00
kwiberg@webrtc.org
0cd5558f2b AudioEncoder subclass for G722 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 11:45:51 +00:00
henrik.lundin@webrtc.org
1db20a4180 Adding EncodedInfo struct to AudioEncoder::Encode 
This struct will be expanded in future changes.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 14:44:50 +00:00
andrew@webrtc.org
1153322cf8 Build fix for MIPS Android Webview build. 
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-21 16:28:32 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
kjellander@webrtc.org
52bb521b47 Update isolate files for Android APK tests. 
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).

A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.

BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 08:35:05 +00:00
turaj@webrtc.org
1431e4dd1c Revert 7675 "Make an AudioEncoder subclass for iSAC" 
Above CL did not compile on Android. Followings are links to Android builds

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320

> Make an AudioEncoder subclass for iSAC
> 
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/25019004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 01:44:13 +00:00
kwiberg@webrtc.org
05feff013e Make an AudioEncoder subclass for iSAC 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 23:53:08 +00:00
kwiberg@webrtc.org
6de75ca3ed Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:29:24 +00:00
kwiberg@webrtc.org
c78cf97ecb Remove the useless dummy state parameter to WebRtcG711_* 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:23:36 +00:00
marpan@webrtc.org
4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff 
Pick up the libvpx roll: https://codereview.chromium.org/674753002

Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3

Clang is not updated in this roll.

Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')

Update rate control parameter in vp9 test.

R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/23229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 20:10:26 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 18:08:09 +00:00
andrew@webrtc.org
508c91683c Build fix for MIPS32R6. 
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 16:26:17 +00:00
kwiberg@webrtc.org
decd9306ae AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket 
Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.

BUG=3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:38:50 +00:00
kwiberg@webrtc.org
663fdd02fd Make an AudioEncoder subclass for Opus 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 07:28:36 +00:00
bjornv@webrtc.org
a296725d0e audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>" 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 13:05:43 +00:00
bjornv@webrtc.org
67ca26e087 common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 
The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t.

Affected components:
* isac/fix

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 13:03:10 +00:00