In some cases, the decoder can write outside of an allocated array. See
the new comment in the code for more details.
BUG=chromium:568885, webrtc:5305
Review URL: https://codereview.webrtc.org/1704463002
Cr-Commit-Position: refs/heads/master@{#11641}
In some cases, the decoder can read outside of an allocated array. See
the new comment in the code for more details.
BUG=chromium:568889, webrtc:5305
Review URL: https://codereview.webrtc.org/1700973002
Cr-Commit-Position: refs/heads/master@{#11637}
Both were related to very large jumps in RTP timestamps.
BUG=webrtc:5488
Review URL: https://codereview.webrtc.org/1685103002
Cr-Commit-Position: refs/heads/master@{#11569}
In some rare occations (very low energy signal), a shift value happened
to be negative. This is now fixed by using the WEBRTC_SPL_SHIFT_W32,
which in essence checks the sign of the number of shifts and performs a
right or left shift accordingly.
The fix reverts to how the code was written in old NetEq; see
4d363ae305/webrtc/modules/audio_coding/neteq/normal.c (165).
BUG=webrtc:5490
Review URL: https://codereview.webrtc.org/1675293002
Cr-Commit-Position: refs/heads/master@{#11546}
This pulls in several fixes and gets Visual Studio 2015 support.
The new repo is located at https://github.com/gflags/gflags
which is mirrored in Chrome infrastructure at
https://chromium.googlesource.com/external/github.com/gflags/gflags
New configuration headers were generated according to README.webrtc
on Windows and Linux. I verified the Linux generated ones are working
on Mac. The generating headers on Mac are identical with only a minor
difference (an __unused attribute) that doesn't effect the build.
BUG=webrtc:5185
NOTRY=True
NOPRESUBMIT=True
TESTED=Successfully ran:
out/Release/video_quality_measurement --input_filename=resources/foreman_cif.yuv --width=352 --height=288
to verify flags are still being parsed properly.
I also ran the compile trybots and the baremetal bots
(since they run tests that have gflags flags).
Review URL: https://codereview.webrtc.org/1679263002
Cr-Commit-Position: refs/heads/master@{#11539}
If a StatisticsCalculator::PeriodicUmaAverage object was created and
then deleted without any samples being logged, the destructor would call
the Metric() method, which calculated sum_/counter_. However, with no
samples logged, counter_ is 0.
This was found and verified using UBSan tests; see the bug for more info.
BUG=webrtc:5490
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1678773003
Cr-Commit-Position: refs/heads/master@{#11534}
This is needed because the target is defined in webrtc/common.gyp
and its current location crosses package boundaries when generating
projects for some build systems.
NOTRY=True
Review URL: https://codereview.webrtc.org/1665603003
Cr-Commit-Position: refs/heads/master@{#11496}
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}
This will make it easier for future CLs to make them optional.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1625363002
Cr-Commit-Position: refs/heads/master@{#11381}
This should have been done in commit 11340, but it was left out by
mistake.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1631443002
Cr-Commit-Position: refs/heads/master@{#11378}
When the file was rewound, the remaining audio read was inserted at
the start of the destination array, not where the first reading
attempt ended.
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1612053002
Cr-Commit-Position: refs/heads/master@{#11343}
Callers can just remember the return value of
RentACodec::RentEncoderStack instead.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1612713002
Cr-Commit-Position: refs/heads/master@{#11340}
Several unittests were disabled on android, this CL will reenable them. One of
the tests was accidentally disabled on all platforms, and now no longer gives a
bitexact result.
BUG=webrtc:3343,webrtc:5349
Review URL: https://codereview.webrtc.org/1532903002
Cr-Commit-Position: refs/heads/master@{#11323}
There's no need for this class to have a vtable since there exists only a single implementation (per platform). It's also not good for performance.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1601743004 .
Cr-Commit-Position: refs/heads/master@{#11306}
That these declarations were missing was a bug, which apparently
didn't actually cause build problems in either Chromium or WebRTC
standalone. (Presumably, because rent_a_codec was always linked
together with other build targets that did declare such dependencies.)
BUG=webrtc:5435
Review URL: https://codereview.webrtc.org/1607463002
Cr-Commit-Position: refs/heads/master@{#11303}
* 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}
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq.
The new RTP file is generated by the following steps:
1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1
2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.)
BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output.
Review URL: https://codereview.webrtc.org/1515113002
Cr-Commit-Position: refs/heads/master@{#11113}
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
NetEqNetworkStatistics has been updated some time ago. A bit exactness test in neteq unittests is still using the old NetEqNetworkStatistics.
New neteq4_network_stats.dat generated by running TestBitExactness with flag "genref"
BUG=
Review URL: https://codereview.webrtc.org/1522103002
Cr-Commit-Position: refs/heads/master@{#11052}
All encoders already handle the "Opus-specific" requests sanely (by
failing nicely), so we don't need extra checks to protect them.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1527453005
Cr-Commit-Position: refs/heads/master@{#11051}
So that the two of them sit next to each other at the top level of
AudioCodingModuleImpl. CodecManager now manages the specifications for
Rent-A-Codec, rather than managing encoders directly.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1520283006
Cr-Commit-Position: refs/heads/master@{#11048}
We already had a special case for android, but it only worked for arm32.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1512833003
Cr-Commit-Position: refs/heads/master@{#10989}
As a step toward fixing webrtc:3987, here we update the RTPencode to allow Opus RTP payloads.
BUG=webrtc:3987, webrtc:2692
Review URL: https://codereview.webrtc.org/1516653003
Cr-Commit-Position: refs/heads/master@{#10987}
By reducing the length of the audio input, the total runtime of
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.*
is reduced by more than 10x, when run single-threaded.
The PCMFile helper class is extended with a FastForward method (to
skip initial silence in the test files) and a limiter on how much to
read.
BUG=webrtc:2463
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1513223002 .
Cr-Commit-Position: refs/heads/master@{#10973}
They were meant to be run if we have either iSAC float or fix, but the
typo made them run for just float.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1513483005
Cr-Commit-Position: refs/heads/master@{#10969}
The new fields are default-populated for built-in decoders, but for
external decoders, the name can now be given when registering the
decoder.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1484343003
Cr-Commit-Position: refs/heads/master@{#10952}
This applies to AcmSwitchingOutputFrequencyOldApi.*,
AcmReceiverBitExactnessOldApi.* and AcmSenderBitExactnessOldApi.*.
BUG=webrtc:4647
NOTRY=true
Review URL: https://codereview.webrtc.org/1503043003
Cr-Commit-Position: refs/heads/master@{#10936}
Also removes virtual from VideoDecoder::Decode and updated mocks and
tests accordingly to use VideoDecoder::DecodeInternal instead.
BUG=webrtc:5167
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1512483003 .
Cr-Commit-Position: refs/heads/master@{#10935}
The bug hasn't caused us any problems, since we don't run CNG together with Opus (our only real 48 kHz codec), but would cause problems if used with PCB16b @ 48 kHz.
BUG=webrtc:5303
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1496243002 .
Cr-Commit-Position: refs/heads/master@{#10929}
