Commit Graph

240 Commits

Author SHA1 Message Date
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
2b1b7a83ad iSAC fix: Ignore overflow in signed left shift
A left shift by 10 was assumed to never overflow, since "[s]imulation
of the 25 files shows that maximum value in the vector gain_lo_hiQ17[]
is 441344, which means that it is log2((2^31)/441344) = 12.2 shifting
bits from saturation." However, a fuzzer test succeeded in provoking
an overflow, which we ignore in this CL on the theory that only
"abnormal" inputs cause overflow.

Also had to replace a "foo << 1" with "foo * (1 << 1)" in
WEBRTC_SPL_MUL_16_32_RSFT15 because foo could be negative; this
problem showed up as soon as I'd asked UBSan to ignore the overflow
discussed above.

BUG=chromium:615819

Review-Url: https://codereview.webrtc.org/2314413002
Cr-Commit-Position: refs/heads/master@{#14162}
2016-09-09 12:51:38 +00:00
d52bef7d64 iSAC float: Handle errors in upper band decoding
We hit a fuzzer bug that caused numDecodedBytesLB + numDecodedBytesUB
> lenEncodedBytes, which is obviously bogus. Check for that, and for
the case whhere the UB decoder itself realized that something was
wrong. (The code already makes the corresponding check for the LB
decoder.)

BUG=chromium:637899

Review-Url: https://codereview.webrtc.org/2315693002
Cr-Commit-Position: refs/heads/master@{#14091}
2016-09-06 13:16:09 +00:00
cfee215b23 Migrated ILBC and ISAC test targets for GN.
Migrated GN targets ilbc_test, isac_api_test,
isac_switch_samprate_test from webrtc/modules/audio_coding/codecs

NOTRY=True
NOPRESUBMIT=True

BUG=webrtc:6191

Review-Url: https://codereview.webrtc.org/2270403002
Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 11:09:28 +00:00
1e8ed4a801 Replace calls to assert() with RTC_DCHECK_*() in .c code
We have RTC_CHECK and RTC_DCHECK for C now, so we should use it. It's
one fewer difference between our C and C++ code.

NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2274083002
Cr-Commit-Position: refs/heads/master@{#13930}
2016-08-26 11:33:41 +00:00
4805231613 Moved format_macros.h from rtc_base to rtc_base_approved.
BUG=webrtc:3806
NOTRY=True

Review-Url: https://codereview.webrtc.org/2272003002
Cr-Commit-Position: refs/heads/master@{#13921}
2016-08-25 11:43:52 +00:00
e51b41ae44 Added GN target for isac_test.
BUG=webrtc:6191
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2267423002
Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 09:26:04 +00:00
fc3a8ee47b Delete unused code.
* Unused audio_coding and video_coding test code.
* Obsolete voice_engine android test app.
* Left-over placeholder files for remoteaudiotrack and
  portallocatorfactory.

In addition, change modules.gyp dependency from rtc_base to
rtc_base_approved.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2065353002 .

Cr-Commit-Position: refs/heads/master@{#13166}
2016-06-16 13:51:40 +00:00
6808419068 iSAC decoder: Remove obsolete TODO
NOTRY=true

Review-Url: https://codereview.webrtc.org/2069993002
Cr-Commit-Position: refs/heads/master@{#13152}
2016-06-15 12:54:20 +00:00
7bf939c720 Check for out-of-bounds access on |kIntrpCoef|.
BUG=chromium:600953

Review-Url: https://codereview.webrtc.org/2025493002
Cr-Commit-Position: refs/heads/master@{#13039}
2016-06-03 17:06:05 +00:00
abe95ba323 AudioDecoderIsacT: Require caller to always specify sample rate
This gets rid of the complex & icky state where the sample rate is not
yet determined.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2020353003
Cr-Commit-Position: refs/heads/master@{#13011}
2016-06-02 09:59:02 +00:00
6c2eab34f8 AudioDecoder: New method SampleRateHz, + implementations for our codecs
This will let NetEq (and the factory, and every layer in between) keep
track of just the decoder, instead of decoder and sample rate.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2024633002
Cr-Commit-Position: refs/heads/master@{#12968}
2016-05-31 09:46:28 +00:00
64208e5523 Fix UBSan errors (left shift of negative value)
I've settled on replacing x << n with x * (1 << n); this gets rid of
the "left shift of negative value" warning, but will still trigger
undefined behavior if the multiplication overflows. It also still
looks like a left shift, which is good for the readability of the
fixed-point code.

(The compiler is smart enough to recognize that the
multiplication+shift is just a shift, for both variable and constant
shift amounts, so the generated code should not change.)

BUG=chromium:603491

Review-Url: https://codereview.webrtc.org/1989803002
Cr-Commit-Position: refs/heads/master@{#12845}
2016-05-23 10:28:35 +00:00
89f237cedc Fix UBSan errors (left shift of negative value, left shift overflows int)
BUG=chromium:603498

Review-Url: https://codereview.webrtc.org/1979973003
Cr-Commit-Position: refs/heads/master@{#12787}
2016-05-18 09:20:35 +00:00
e305d956c0 Remove runtime NEON detection
Chrome does not detect NEON instruction set at runtime in WebRTC code starting
with M50, which is now in Stable. Remove support for runtime detection for
simplicity.

The only remaining piece of Chrome that will continue to depend on runtime
detection is /net, where devices with _broken_ neon support are also detected,
and it is not configurable via GYP/GN.

BUG=522035
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/1955413003
Cr-Commit-Position: refs/heads/master@{#12778}
2016-05-17 17:56:48 +00:00
8bce67b745 Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created.
Review-Url: https://codereview.webrtc.org/1967503002
Cr-Commit-Position: refs/heads/master@{#12760}
2016-05-16 14:34:31 +00:00
ef8b61e110 Enable -Winconsistent-missing-override flag.
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.

NOPRESUBMIT=True
BUG=webrtc:3970

Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
2016-04-29 13:09:23 +00:00
4485ffb58d #include "webrtc/base/constructormagic.h" where appropriate
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.

Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1917043005

Cr-Commit-Position: refs/heads/master@{#12509}
2016-04-26 15:14:48 +00:00
2903ba5ff3 Reland Remove the deprecated EncodeInternal interface from AudioEncoder
Remove the deprecated EncodeInternal interface from AudioEncoder

Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.

BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1881003003

Cr-Commit-Position: refs/heads/master@{#12409}
2016-04-18 13:14:42 +00:00
164bc4bbd3 Revert of Remove the deprecated EncodeInternal interface from AudioEncoder (patchset #4 id:60001 of https://codereview.webrtc.org/1864993002/ )
Reason for revert:
Broke import. Implementations of the old interface still exists somewhere.

Original issue's description:
> Remove the deprecated EncodeInternal interface from AudioEncoder
>
> Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
>
> BUG=webrtc:5591
>
> Committed: https://crrev.com/5222d315dbea8f3563c100cc9f2451907f70b05f
> Cr-Commit-Position: refs/heads/master@{#12329}

TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1883543002

Cr-Commit-Position: refs/heads/master@{#12330}
2016-04-12 10:58:10 +00:00
5222d315db Remove the deprecated EncodeInternal interface from AudioEncoder
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.

BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1864993002

Cr-Commit-Position: refs/heads/master@{#12329}
2016-04-12 10:31:03 +00:00
0d05da7ee6 Rent-A-Codec: Reference count the shared iSAC bandwidth estimation state
Now that the Rent-A-Codec no longer owns the encoders and decoders it
produces, they may outlive it. It's thus no longer correct for the
Rent-A-Codec to own the bandwidth estimation state; all of the
involved objects need to share ownership.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1821513003

Cr-Commit-Position: refs/heads/master@{#12159}
2016-03-30 11:10:18 +00:00
15622c0aaf WebRtcIsacfix_PitchFilter: Don't read uninitialized array entries
WebRtcIsacfix_PitchFilterCore requires indW32 >= PITCH_FRACORDER - 2;
otherwise, it will read from entries of ubufQQ that haven't been
written yet. (The problem of indW32 being too small has only been seen
in fuzzer tests, not in real life.)

BUG=chromium:581901

Review URL: https://codereview.webrtc.org/1811453002

Cr-Commit-Position: refs/heads/master@{#12047}
2016-03-18 05:17:21 +00:00
4f43fcfd49 Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.
Renamed the new variant of EncodeInternal to EncodeImpl, so that
subclasses implementing one of the EncodeInternal don't need to
explicitly contain 'using AudioEncoder::EncodeInternal' to avoid their
implementation hiding the other variant of EncodeInternal. This causes
a warning (treated as an error) when building using GCC.

Review URL: https://codereview.webrtc.org/1764583003

Cr-Commit-Position: refs/heads/master@{#11868}
2016-03-04 08:54:38 +00:00
10a029e952 Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.
For backwards compatibility, I've added kept the old interface to
Encode() and EncodeInternal and created default implementations of both
variants of EncodeInternal(), each calling the other. At least one of
the variants must be implemented in a subclass or we'll run out of stack
and explode. Would be nice if we could catch that before runtime. :/

The new interface to EncodeInternal() is protected, since it should
never be called from the outside.

Was unable to mark the old EncodeInternal() as RTC_DEPRECATED, since the
default implementaion of the new variant needs to call it to work around
old implementations. The old Encode() variant is deprecated, at least.

Added a test for backwards compatibility in audio_encoder_unittest.cc.
For the added test I broke out MockEncodeHelper from
audio_encoder_copy_red_unittest.cc and renamed it MockAudioEncoderHelper.

Review URL: https://codereview.webrtc.org/1725143003

Cr-Commit-Position: refs/heads/master@{#11823}
2016-03-01 08:41:39 +00:00
806706875d iSAC entropy coder: Avoid signed integer overflow
By doing an unsigned instead of a signed addition, we get the exact
same machine code (in non-UBSan builds), but no longer trigger
undefined behavior since unsigned overflow is defined behavior.

BUG=webrtc:5485

Review URL: https://codereview.webrtc.org/1734883003

Cr-Commit-Position: refs/heads/master@{#11776}
2016-02-26 10:52:14 +00:00
b7261fd3ae iSAC float: Check for end of input buffer while decoding
Previously, we relied on the encoded stream to come to an end before
the end of the buffer. This is a bad idea, since it is possible to
craft a stream that fills the buffer while decoding to less than the
expected amount of data; without the new checks introduced here, this
causes the decoder to read past the end of the input buffer.

BUG=chromium:582471, chromium:587852

Review URL: https://codereview.webrtc.org/1721593004

Cr-Commit-Position: refs/heads/master@{#11734}
2016-02-24 09:34:33 +00:00
91d9756bc6 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1696853004

Cr-Commit-Position: refs/heads/master@{#11613}
2016-02-14 09:10:08 +00:00
3f70562bbb Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015).
Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast.

Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000

Hiding snprintf definition if building with Visual Studio 2015

Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc.

BUG=webrtc:5183

Review URL: https://codereview.webrtc.org/1412653006

Cr-Commit-Position: refs/heads/master@{#11434}
2016-01-30 22:40:52 +00:00
5ad935cb56 Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1613643004

Cr-Commit-Position: refs/heads/master@{#11367}
2016-01-25 11:52:53 +00:00
9090e0b147 Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding.
This is a part of cleaning up CriticalSectionWrapper in general.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1610073003 .

Cr-Commit-Position: refs/heads/master@{#11319}
2016-01-20 12:39:45 +00:00
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
25702cb162 Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08 21:50:32 +00:00
740c367af3 iSAC: Remove unnecessary WEBRTC_LINUX define.
I can only find one use in iSAC codebase:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc/modules/audio_coding/test/iSACTest.cc&l=19

It's the prime suspect for causing a compilation error for iOS failing to
include linux/net.h which is being included in
webrtc/voice_engine/voice_engine_defines.h

NOTRY=True

Review URL: https://codereview.webrtc.org/1539883002

Cr-Commit-Position: refs/heads/master@{#11089}
2015-12-18 20:28:28 +00:00
7cae30cbe1 Disable warnings failing when using Clang on Windows.
This makes it possible to build WebRTC using Clang on Windows.
Depends on https://codereview.webrtc.org/1524703006/

BUG=webrtc:5360, webrtc:5366
NOTRY=True

Review URL: https://codereview.webrtc.org/1522223002

Cr-Commit-Position: refs/heads/master@{#11058}
2015-12-16 22:05:36 +00:00
288886b2ec Pass audio to AudioEncoder::Encode() in an ArrayView
Instead of in separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1418423010

Cr-Commit-Position: refs/heads/master@{#10535}
2015-11-06 09:21:39 +00:00
a41ab9326c Switch usage of _DEBUG macro to NDEBUG.
http://stackoverflow.com/a/29253284/5237416

BUG=None
R=tommi@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1429513004

Cr-Commit-Position: refs/heads/master@{#10468}
2015-10-30 23:08:54 +00:00
74640895fa audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417173004 .

Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 10:31:11 +00:00
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
5460f9b81d Workaround for false positive -Wmaybe-uninitialized being triggered on some compilers
Some toolchains (in this case referring to a g++ 4.9, or "arm-linux-
androideabi-g++ (GCC) 4.9 20140827 (prerelease)" according to my
--version, from the Android NDK r10e-rc4 and potentially with custom
patches; others may be affected as well) fail to prove that myVec in
WebRtcIsac_CorrelateInterVec is never used uninitialized. This is likely
due to the compiler thinking the assignment in line 468 might not
happen. Changing the loop condition in line 466 to rowCntr <
SOME_CONSTANT also helps, suggesting that the compiler can't infer that
there are only 2 values interVecDim can have at that point, and neither
of them are 0. Of course, this is not an acceptable fix, as it changes
behaviour.

This seems to be a compiler bug, or at least an issue with its
heuristics. However, we can't really change toolchains at the moment,
and ultimately this change improves support for certain older compilers.

BUG=

Review URL: https://codereview.webrtc.org/1406423004

Cr-Commit-Position: refs/heads/master@{#10337}
2015-10-20 12:45:09 +00:00
bd7de0c6ef Delete full-band mode from the iSAC codec
This mode is no longer used.

BUG=4210

Review URL: https://codereview.webrtc.org/1392173004

Cr-Commit-Position: refs/heads/master@{#10275}
2015-10-14 13:06:00 +00:00
8967183bf7 Simple cleanups of AudioDecoder and AudioEncoder classes
* Make sure they're all final and don't allow copying or assignment.

  * Get rid of the single-channel PCM decoder classes.

  * Move some includes from .h to .cc files where possible.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1353803002

Cr-Commit-Position: refs/heads/master@{#10021}
2015-09-22 21:06:34 +00:00
7404368998 Move AudioDecoderIsac* to its own files
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1339253003 .

Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22 17:31:52 +00:00
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
3c089d751e Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +00:00
c99ebc1490 Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize
And the corresponding ACM methods SetISACMaxRate and
SetISACMaxPayloadSize. They were only used in tests.

Review URL: https://codereview.webrtc.org/1311533010

Cr-Commit-Position: refs/heads/master@{#9903}
2015-09-09 07:54:10 +00:00
12cfc9b4da Fold AudioEncoderMutable into AudioEncoder
It makes more sense to combine the two interfaces, since there wasn't
a clear line separating them. The result is a combined interface with
just over a dozen methods, half of which need to be implemented by
every subclass, while the other half have sensible (and trivial)
default implementations and are implemented only by the few subclasses
that need non-default behavior.

Review URL: https://codereview.webrtc.org/1322973004

Cr-Commit-Position: refs/heads/master@{#9894}
2015-09-08 12:57:59 +00:00
b7306ae6fe Revert "Avoiding size_t in MIPS."
This reverts commit 32e2f461b13c530d34f9c434e7e76da6ff3eda83.

BUG=526716
TBR=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1314873004 .

Cr-Commit-Position: refs/heads/master@{#9824}
2015-08-31 18:44:05 +00:00
32e2f461b1 Avoiding size_t in MIPS.
TBR=pkasting@chromium.org

BUG=526716

Review URL: https://codereview.webrtc.org/1310553008 .

Cr-Commit-Position: refs/heads/master@{#9823}
2015-08-31 15:40:17 +00:00