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}
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}
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}
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}
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}
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}
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/isac/ are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1179093002
Cr-Commit-Position: refs/heads/master@{#9422}
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question. This is preparation for a future change
that will convert a variety of types to size_t.
There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.
BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm
Review URL: https://codereview.webrtc.org/1174813003
Cr-Commit-Position: refs/heads/master@{#9413}
This primarily addresses two things:
* Tab characters still present, mostly in comments
* printfs split across multiple lines in a suboptimal way
Along the way this fixes a few spelling errors and other minor changes.
BUG=none
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52689004
Cr-Commit-Position: refs/heads/master@{#9406}
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54629004
Cr-Commit-Position: refs/heads/master@{#9405}
The existing style in these files is pretty inconsistent and wildly divergent
from most of WebRTC/Chromium; clang-formatting them not only makes them easier
to read, it makes me see fewer presubmit errors when I try to touch the files to
make other changes.
BUG=none
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52019004
Cr-Commit-Position: refs/heads/master@{#9364}
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44659004
Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.
The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.
BUG=4361
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45619004
Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
The current way that iSAC RCU is packetized and sent as a RED packet,
with the same payload type for primary and redundant payloads, does
not follow the specification for RED. As it is now, it is impossible
for a receiver to know if an incoming RED packet with iSAC payloads
inside consists of two "primary" (but time-shifted) payloads, or one
primary and one RCU payload. The RED standard stipulates that the
former option is the correct interpretation, while our implementation
currently applies the latter.
This CL removes support for iSAC RCU from Audio Coding Module, but
leaves it in the iSAC codec itself (i.e., in the C implementation).
BUG=4402
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45569004
Cr-Commit-Position: refs/heads/master@{#8713}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8713 4adac7df-926f-26a2-2b94-8c16560cd09d
Added method AudioEncoder::MaxEncodedBytes() and provided implementations in derived encoders. This method returns the number of bytes that can be produced by the encoder at each Encode() call.
Unit tests were updated to use the new method.
Buffer allocation was not changed in AudioCodingModuleImpl::Encode(). It will be done after additional investigation.
Other refactoring work that was done, that may not be obvious why:
1. Moved some code into AudioEncoderCng::EncodePassive() to make it more consistent with EncodeActive().
2. Changed the order of NumChannels() and RtpTimestampRateHz() declarations in AudioEncoderG722 and AudioEncoderCopyRed classes. It just bothered me that the order was not the same as in AudioEncoder class and its other derived classes.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40259005
Cr-Commit-Position: refs/heads/master@{#8671}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8671 4adac7df-926f-26a2-2b94-8c16560cd09d
Without this patch, Valgrind's Memcheck was complaining that the test
for whether we should return -1 following the call to
WebRtcIsac_DecodeLb made a conditional branch or move based on the
value of numSamplesLB, which was uninitialized if WebRtcIsac_DecodeLb
failed.
However, as can be seen in the source, the control flow only depends
on the value of numSamplesLB if numDecodedBytesLB >= 0; i.e., if
WebRtcIsac_DecodeLb returned successfully, in which case numSamplesLB
is always initialized. The discrepancy is due to the fact that
Valgrind works on the generated machine code, which contains spurious
such dependencies. The generated code for this test:
if ((numDecodedBytesLB < 0) || (numDecodedBytesLB > lenEncodedLBBytes) ||
(numSamplesLB > MAX_FRAMESAMPLES)) {
looks like this:
95: 0f bf 45 d6 movswl -0x2a(%rbp),%eax
99: 3d c0 03 00 00 cmp $0x3c0,%eax
9e: 0f 8f 45 01 00 00 jg 1e9 <Decode+0x1e9>
a4: 44 89 f0 mov %r14d,%eax
a7: c1 e0 10 shl $0x10,%eax
aa: 0f 88 39 01 00 00 js 1e9 <Decode+0x1e9>
b0: 41 0f bf ce movswl %r14w,%ecx
b4: 89 8d 98 e1 ff ff mov %ecx,-0x1e68(%rbp)
ba: 41 0f bf c7 movswl %r15w,%eax
be: 39 c1 cmp %eax,%ecx
c0: 0f 8f 23 01 00 00 jg 1e9 <Decode+0x1e9>
Note how the compiler has seemingly ignored the C language's guarantee
that the arguments to || must be evaluated in left-to-right order, and
compares numSamplesLB (%eax) with MAX_FRAMESAMPLES (0x3c0, a.k.a. 960)
before the other two conditions! If the uninitialized value in
numSamplesLB happens to be greater than 960, we'll jump to
Decode+0x1e9 (where we'll return -1) without even looking at the other
two conditions. Has the compiler generated broken code?
Well, no. If numDecodedBytesLB is < 0 so that numSamplesLB is
uninitialized, we'll end up jumping to 1e9 whether that value is
greater than 960 or not; we'll just do it with different jump
instructions. This is entirely invisible as far as the C language is
concerned, but the dependency on the uninitialized value is visible at
the machine code level, which is why Memcheck complains.
This patch solves the problem by pragmatically initializing
numSamplesLB before the call even though it isn't necessary other than
for placating Memcheck.
BUG=4143
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36309004
Cr-Commit-Position: refs/heads/master@{#8492}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8492 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add max_bit_rate and max_payload_size_bytes to config structs.
- Fix support for 48 kHz sample rate.
- Fix iSAC-RED.
- Add method UpdateDecoderSampleRate().
- Update locking structure with a separate lock for local member
variables used by the encoder methods.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41659004
Cr-Commit-Position: refs/heads/master@{#8204}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8204 4adac7df-926f-26a2-2b94-8c16560cd09d
