This is no longer used. Related code in NetEq and the iSAC codec itself
will be deleted in follow-up CLs.
BUG=4210
Review URL: https://codereview.webrtc.org/1404623002
Cr-Commit-Position: refs/heads/master@{#10264}
* Last major but partial merge from upstream
was from r2699 on 9/17/2012.
Manual changes after "git merge 9b1cf54"
* Remove LOCAL_CLANG:=false
* Fix or workaround clang compilation error:
* Remove unnecessary LOCAL_CLANG_* flags or add comment.
* Fix type cast error and asm code error in
src/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
* Keep AOSP's Android.mk files, but update include path,
local source file lists, and new -D predefined macros.
* Remove new Android.mk files from r2699.
We do not need new features from r2699.
* Remove all definition and use of WEBRTC_TARGET_PC and WEBRTC_USE_SSE2.
They are not needed with new upgraded Android.mk files.
* Keep old AOSP patches to compile on Android in
src/system_wrappers/source/android/cpu-features.c
src/modules/audio_coding/codecs/isac/fix/test/kenny.c
src/common_audio/signal_processing/spl_sqrt_floor.c
src/typedefs.h
* In /src/common_audio/signal_processing, the following C files have
new definitions from r2699, and some .s files for neon.
The new .s files have compilation error on AOSP, so we skip them.
The .s files are changed to .S in newer upstream versions that
we could merge in later and hopefully resolve all the problems.
The new .c files should also work, no worse than current AOSP .c files.
min_mnax_operations_neon.c
cross_correlation.c
downsample_fast.c
filter_ar_fast_q12.c
complex_bit_reverse.c
* Use nsx_core_neon.c instead of nsx_core_neon.S for arm64 target
because of arm64 compilation errors on AOSP.
* Remove unused parameter in src/modules/interface/module.h.
* Remove ./test, src/.../Testsort.cpp, and other files that were in
AOSP but not used and not in r2699.
* Some r2699 features are not ported to AOSP:
MY_WEBRTC_ROOT_PATH, LOCAL_PRELINK_MODULE, libstlport,
TARGET_SIMULATOR, webrtc_audioproc, webrtc_audioproc_unittest,
external/gtest/include, libgtest, libstlport
* Remove WEBRTC_USE_SSE2, no need after upgrade.
* Copy upstream *.gypi files, but they are not used in AOSP build.
After this merge of r2699, we should have git merge history
to simplify future merges.
Change-Id: I703e59bdb27d60cd5aab9f0744dd8ce25d0ea339
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.
The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:
void Foo::Bar() const {}
rtc::Callback0<void> Foo::MakeClosure() const {
return rtc::Bind(&Foo::Bar, this);
}
We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.
A test is also added in bind_unittest to check this behaviour.
BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1403683004 .
Cr-Commit-Position: refs/heads/master@{#10253}
Reason for revert:
Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works.
Original issue's description:
> Adding support for simulcast and spatial layers into VideoQualityTest
>
> The CL includes several changes:
> - Adding flags describing the streams and spatial layers.
> - Reorganizing the order of the flags, to make them easier to maintain.
> - Adding a member .params_ to VideoQualityAnalyzer.
> (instead of passing it to every member function manually)
> - Updating VideoAnalyzer to support simulcast.
> (select appropriate ssrc and fix timestamps which are sometimes increased by 1)
> - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
> Changing to first read bitrates and resolution ratios from the flags, if specified.
> If not specified, reverting to the old code are setting the values automatically.
> - Changing the parameters in LayerFilteringTransport, replacing
> xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
>
> Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e
> Cr-Commit-Position: refs/heads/master@{#10215}
TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1397363002
Cr-Commit-Position: refs/heads/master@{#10252}
This make small refactorings to MediaVideoEncoder to prepare for adding support to encode from textures. The C++ layer does not have any functional changes.
- Moves ResetEncoder to always work on the codec thread
- Adds use of ThreadChecker.
- Change Java MediaEncoder.Init to return true or false and introduce method getInputBuffers.
- Add simple unit test for Java MediaCodecVideoEncoder.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1396073003
Cr-Commit-Position: refs/heads/master@{#10250}
This CL is a small bug fix for "Android SurfaceViewRenderer: Allow to re-init after release() has been called" https://codereview.webrtc.org/1389203003/. It is only possible to clear the last image in release() if init() has been called beforehand.
TBR=hbos
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1396573003 .
Cr-Commit-Position: refs/heads/master@{#10223}
This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.
This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.
BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1348113003 .
Cr-Commit-Position: refs/heads/master@{#10221}
These are the necessary changes in C++ related to the video capturer necessary to capture to a surface texture.
It does not handle scaling / cropping yet though.
BUG=
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1395673003 .
Cr-Commit-Position: refs/heads/master@{#10218}
This CL makes a thorough reset of all variables in release() and clears the last rendered image so that the SurfaceViewRenderer object can be reinitialized with init() and work properly. This CL also removes an implicit assumption that init() is called before surfaceCreated() - now they can be called in any order.
BUG=webrtc:4742
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1389203003 .
Cr-Commit-Position: refs/heads/master@{#10217}
What used to be the libpeerconnection library is now compiled
statically into the Chromium binary, so clean up references it.
BUG=chromium:482123
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1399513002 .
Cr-Commit-Position: refs/heads/master@{#10216}
