* 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}
Also clean up some include_dir entries and update the few
references to them with absolute include paths instead.
Finally fixed a few lint errors and invalid header guards.
None of these are used downstream.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1438663003 .
Cr-Commit-Position: refs/heads/master@{#10700}
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself.
Review URL: https://codereview.webrtc.org/1446093003
Cr-Commit-Position: refs/heads/master@{#10653}
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
BUG=webrtc:5127
Review URL: https://codereview.webrtc.org/1415173005
Cr-Commit-Position: refs/heads/master@{#10565}
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,
2. NetEq DoCodecInternalCng did not assign enough buffer.
P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.
BUG=webrtc:4985
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1334303005 .
Cr-Commit-Position: refs/heads/master@{#10031}
* 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}
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}
Fix bug 4981, which caused the second half (decreasing loss rates) to
not test anything. In the process, the test is changed slightly to
make it less dependent on the exact rounding behavior of doubles (by
not testing exactly at the the points where the effective loss rate
goes through a step---just very very close). A bunch of symbolic
constants are also replaced with easy-to-read literal numbers.
BUG=4981
Review URL: https://codereview.webrtc.org/1316673010
Cr-Commit-Position: refs/heads/master@{#9908}
Merge the contents of audio_encoder_mutable_opus_test.cc into
audio_encoder_opus_unittest.cc, since they're now both testing
AudioEncoderOpus.
(While preparing this CL, I noted a bug in the PacketLossRateOptimized
test. This CL leaves that test essentially unchanged; I've posted bug
4981 about the problem.)
Review URL: https://codereview.webrtc.org/1319713004
Cr-Commit-Position: refs/heads/master@{#9906}
There's no point in returning a status code, since the max playback rate
is only a suggestion that the encoder is free to disregard.
Review URL: https://codereview.webrtc.org/1332573003
Cr-Commit-Position: refs/heads/master@{#9900}
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}
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}
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}
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/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, 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/1179093003
Cr-Commit-Position: refs/heads/master@{#9424}
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}
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}
A better solution than forcing OPUS_APPLICATION_VOIP when enabling DTX has been found, which is to set OPUS_SIGNAL_VOICE.
This reduces the uncertainty of entering DTX over silence period of audio.
This CL contains the setup of OPUS_SIGNAL_VOICE and decoupling opus application mode with DTX.
BUG=4559
R=henrik.lundin@webrtc.org, henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46959004
Cr-Commit-Position: refs/heads/master@{#9168}
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
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
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
This solution does not use the existing VAD/DTX logic of ACM, since Opus DTX is codec feature, while ACM VAD/DTX is mainly for setting the WebRTC VAD/DTX.
During the development of this CL, two old bugs were found and are fixed in this CL too.
They are in
webrtc/modules/audio_coding/test/Channels.cc
and webrtc/modules/audio_coding/main/acm2/acm_opus_unittest.cc
respectively.
BUG=webrtc:1014
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38469004
Cr-Commit-Position: refs/heads/master@{#8573}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8573 4adac7df-926f-26a2-2b94-8c16560cd09d
The old AcmOpusTest depends on the ACMOpus class, but this class was
obsoleted by AudioEncoderOpus. In this CL, the test code is re-written
to use AudioEncoderOpus and ACMGenericCodecWrapper instead of
ACMOpus. Most of the test functionality is preserved, except for the
packet loss rate tests, which where already transferred to
AudioEncoderOpusTest in r8244.
R=kwiberg@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40029004
Cr-Commit-Position: refs/heads/master@{#8410}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8410 4adac7df-926f-26a2-2b94-8c16560cd09d
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz()
are simple accessors for almost all implementations of AudioEncoder,
they are virtual and not guaranteed to be just simple accessors. Thus,
it makes more sense to use the normal CamelCase naming scheme.
BUG=4235
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34239004
Cr-Commit-Position: refs/heads/master@{#8407}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
