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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
* 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}
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}
* 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}
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}
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}
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}
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}
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}
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.
Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1319683002 .
Cr-Commit-Position: refs/heads/master@{#9798}
The only shared state is now the bandwidth estimation info.
This reduces the amount and complexity of the locking
substantially.
Review URL: https://codereview.webrtc.org/1208993010
Cr-Commit-Position: refs/heads/master@{#9762}
WebRtcIsacfix_AllpassFilter2FixDec16Neon was disabled due to a Clang
bug. The bug is fixed in current Clang version, re-enable it in this patch.
BUG=4567
R=andrew@webrtc.org, kjellander@webrtc.org
TEST=buildbot build
Change-Id: I71e309cec6caf376181cf9c299c9e8967c9a328e
Review URL: https://codereview.webrtc.org/1194773002 .
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
Cr-Commit-Position: refs/heads/master@{#9645}
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
This patch tests separate iSAC encoder and decoder in more cases (32
kHz in addition to 16 kHz, and 30 ms adaptive and 60 ms nonadaptive).
In order to handle 32 kHz adaptive, the decoder needs to be told of
the encoder's sample rate (16 kHz worked already because that's the
default). And since we can't set the encoder's frame size without also
setting its bit rate, we need a way to set the decoder's bit rate as
well.
It turned out to be way too messy to continue verifying that the
bandwidth estimator does something reasonable in all these cases,
because it seems it doesn't. So the GetSetBandwidthInfo is now just
responsible for ensuring that split encoder/decoder behaves the same
as conjoined encoder/decoder; the job of verifying that the bandwidth
estimator does its job properly falls on some other test (that doesn't
exist yet).
Review URL: https://codereview.webrtc.org/1225093005
Cr-Commit-Position: refs/heads/master@{#9583}
They make it possible to send bandwidth estimation info from decoder
to encoder even if they are separate objects (which we want them to be
because multithreading).
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1208923002.
Cr-Commit-Position: refs/heads/master@{#9535}
Using random "garbage" bytes makes testing harder for no good reason.
Any deterministic sequence would do, but we choose all zeros because
it's simple.
Review URL: https://codereview.webrtc.org/1211243014
Cr-Commit-Position: refs/heads/master@{#9532}
The GetTargetBitrate implementation will return the
target bitrate of the codec. This may differ from the
desired target bitrate, as set by SetTargetBitrate, depending on implementation.
Tests are updated to exercise the new functionality.
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1184313002.
Cr-Commit-Position: refs/heads/master@{#9461}
These tables are constant, so it makes sense for all encoders to share
one copy---but it was initialized in a racy way, and there's no
appealing way to fix that without adding dependencies on locking
functions. So we simply give each codec instance its own copy, which
costs 8 * (240 + 240 + 120 + 120) = 5760 bytes apiece.
As noted in the TODO comment, the size of the tables could be reduced,
and they could be filled in at compile-time, but that would make the
encoder output slightly different, which would mess with our tests.
R=henrik.lundin@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1177993003.
Cr-Commit-Position: refs/heads/master@{#9442}
Before this change, it could happen that a caller would get a pointer
to the encoder_ but not use it before another thread called the
Reconstruct method, changing the pointer. This of course resulted in
bad access crashes. With this change, each use of the pointer acquired
from the encoder() method is protected by the same lock that is
required to update the pointer. Note that this fix is probably too
aggressive, since it also affects the Opus implementation; the crash
has so far only been seen for iSAC.
Also adding a test to trigger the problem. The test did not trigger
the problem deterministically, but out would typically find it in less
than 1000 runs.
BUG=chromium:499468
R=jmarusic@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1176303004.
Cr-Commit-Position: refs/heads/master@{#9436}
