It doesn't work, and only complicates the Soong conversion (since it
can't even be tested). It shouldn't be that hard to reimplement in Soong
if we ever fix it.
Test: mmma external/webrtc
Change-Id: Ia5a23a6b7fb1ee17f7d9633016eb8ff73435b448
* We should enable those warnings after fixing the upstream source code.
Bug: 64487164
Test: build with WITH_TIDY=1
Change-Id: I306b4eb26fc8729f21cab4d5b82c9e3b59d8db3e
Native tests are not moved. Tests include:
webrtc_apm_process_test
webrtc_isac_test
webrtc_apm_unit_test
Test: mma -j
Test: m -j -k BOARD_VNDK_VERSION=current has no errors for webrtc
Bug: 62489821
Merged-In: I013303de263866cbf368f3f89327c5357f9cecdb
Change-Id: I013303de263866cbf368f3f89327c5357f9cecdb
(cherry picked from commit 8df7e85368569b7cd0afc1ce231b8b3a0ab49333)
We should not copy OWNERS files from upstream,
or the owners should be registered in Gerrit Code Review.
Bug: 33166666
Test: default build targets
Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
* git merge 04cb763
* See all upstream changes since the previous merge in branch aosp/upstream-master: git diff cb3f9bd..04cb763
* Modify webrtc/.gitignore to keep *.mk files.
* Removed old files from *.mk files:
- thread.cc
- thread_posix.cc
* Add new files to *.mk files:
- event_tracer.cc
* Android relevant upstream changes:
- Make Beamforming dynamically settable for Android platform builds
- Remove additional channel constraints when Beamforming is enabled in AudioProcessing
- Use an explicit identifier in Config
Change-Id: I384a4e8f6982c31c5bc70eef521bb2d90800b9a4
* 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 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}
Changes after "git merge cb3f9bd"
* git mv old Android.mk from src/ to webrtc/
* Remove old unused files in src/*.
* Modify webrtc/.gitignore to keep *.mk files.
* Copy old files from master, lost in auto-merge.
src/modules/audio_processing/test/unit_test.cc
src/modules/audio_coding/codecs/isac/fix/test/{Android.mk,kenny.c}
to webrtc, but most of the old test code do not compile with new
webrtc API and are commented out.
* Move src/modules/audio_processing/test/android/apmtest/jni/*.mk to
webrtc/... but the Android.mk files does not work.
Commented out its build target.
* Changes to Android.mk files:
* Change references of src/ to webrtc/.
* Fix include path
* Fix source file list, remove old non-existing files,
add new source files to resolve link errors.
* Add new Android.mk files to build some new static libraries
to link into current Android webrtc .so files.
* Remove unnecessary LOCAL_SHARED_LIBRARIES in Android.mk files
that build static libraries.
* Remove old unnecessary clang workarounds like
-Wno-tautological-pointer-compare
-no-integrated-as
* Fix include path of debug.pb.h in some source files.
* Add -DWEBRTC_POSIX in android-webrtc.mk
* Manually merge Android specific changes in
src/typedefs.h to webrtc/typedefs.h
* Fix trivial syntax error in scoped_ptr.h, calling static_assert.
* Use -std=c++0x in webrtc/system_wrappers/source/Android.mk
* #undef getchaar in spreadsort.hpp
* Verified and not to carry old Android hacks from src/... to webrtc/...
src/system_wrappers/source/android/cpu-features.c
src/modules/interface/module.h
src/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
src/system_wrappers/source/trace_posix.cc
src/typedefs.h
More pathes from Alex Luebs:
* Use new unit test kenny.cc.
Delete old kenny.cc.
Comment out unessential code in kenny.cc to fix link error for now.
* Replace old unit test files with new ones in
webrtc/modules/audio_processing/Android.mk.
Delete old audio_processing/test/unit_test.cc.
* Fix compilation errors in
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
Change-Id: I7bbf776eeb9dcfa21a82dd1f2dec378235cbbc3e
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}
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}
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}
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}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348613003
Cr-Commit-Position: refs/heads/master@{#10015}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1346993002
Cr-Commit-Position: refs/heads/master@{#9966}
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}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348053002
Cr-Commit-Position: refs/heads/master@{#9961}
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}
This makes the sequence of expected calls easier to read. Also, we can
save one line and get rid of a gmock warning by expecting the
MockAudioEncoder object to be destroyed at the end of the test instead
of making a final marker call.
Review URL: https://codereview.webrtc.org/1331793003
Cr-Commit-Position: refs/heads/master@{#9916}
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}
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}
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 easy way also happens to be more efficient if we have to
reallocate, but that's a minor concern here.
Review URL: https://codereview.webrtc.org/1327053002
Cr-Commit-Position: refs/heads/master@{#9876}
