This will make it easier to track which revision is
in a certain Chrome release, since you don't have to
look up the Git hash every time.
Also rename svn_revision to commit_position to make
it more clear what the number is in the post-SVN era.
TESTED=tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /ssd/chrome/src --verbose --ignore-checks --dry-run --close-previous-roll
and verified in the modified DEPS file that the comment was set.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49439004
Cr-Commit-Position: refs/heads/master@{#8756}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8756 4adac7df-926f-26a2-2b94-8c16560cd09d
pthread_cond_{timedwait,wait} are allowed to spuriously wake up as if
they were signaled. To prevent this being interpreted as a "real"
signaling of the event (ThreadWrapper for instance depends on it being
an actual signal) we need to check whether the event was actually
signalled or not.
BUG=4413
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49369004
Cr-Commit-Position: refs/heads/master@{#8752}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8752 4adac7df-926f-26a2-2b94-8c16560cd09d
Now when we don't use SwapFrame consistently anymore, we need to recycle allocations with a buffer pool instead. This CL adds a buffer pool class, and updates the vp8 decoder to use it. If this CL lands successfully I will update the other video producers as well.
BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41189004
Cr-Commit-Position: refs/heads/master@{#8748}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8748 4adac7df-926f-26a2-2b94-8c16560cd09d
Pulls in new libvpx version that allows us to re-enable the
VideoProcessorIntegrationTest.ProcessNoLossDenoiserOnVP9
test in webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
Relevant changes:
* src/third_party/libvpx: 763fe7a..f80cf58
* src/tools/gyp: 4a9b712..d174d75
Details: 8d51d96..bd49b12/DEPS
Clang version was not updated in this roll.
BUG=4418
TBR=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41339004
Cr-Commit-Position: refs/heads/master@{#8745}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8745 4adac7df-926f-26a2-2b94-8c16560cd09d
As of r7849 the built-in AEC on devicing supporting it is enabled by default.
Unfortunately, the SW AEC (AECM) was not disabled, hence running on top of the built-in one. This is not necessary. In fact it reduce double talk performance significantly.
BUG=4431
TESTED=manually
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49419004
Cr-Commit-Position: refs/heads/master@{#8735}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8735 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
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: 00e438c..8d51d96/DEPS
This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.
Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.
Clang version was not updated in this roll.
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42779004
Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
The --close-previous-roll makes it possible to always
close a previously created roll when creating a new one.
This way it will be possible to avoid getting a pile of
open CLs created and never closed for all failed
roll attempts, which is useful for automation.
I also moved some variables out of the AutoRoller
class that doesn't neeed to be there.
BUG=chromium:433305
TESTED=Ran:
tools/autoroller/roll_webrtc_in_chromium.py --verbose --close-previous-roll
and verified it actually closed an existing roll.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40349004
Cr-Commit-Position: refs/heads/master@{#8726}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8726 4adac7df-926f-26a2-2b94-8c16560cd09d
On fresh checkout AppRTCDemo and corresponding tests
fail to build because resource file R.java is not auto generated properly.
On existing tree R.java will be picked up from previous
build leftover at talk/examples/android/gen.
Build bots did not detect this break for some reason.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43749004
Cr-Commit-Position: refs/heads/master@{#8723}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8723 4adac7df-926f-26a2-2b94-8c16560cd09d
Add a --wait-for-trybots flag that will have the script look for
a previously created roll and poll the trybot status until at least
one trybot has failed or all have compelted successfully.
BUG=chromium:433305
TESTED=Ran:
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /ssd/chrome/src --verbose
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /ssd/chrome/src --verbose --wait-for-trybots
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45659004
Cr-Commit-Position: refs/heads/master@{#8718}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8718 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see
common_audio/signal_processing/include/spl_inl_armv7.h and
common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int
or int32_t)
Some other minor code cleanup also exists.
BUG=3348,3353
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42639004
Cr-Commit-Position: refs/heads/master@{#8717}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8717 4adac7df-926f-26a2-2b94-8c16560cd09d
Send the created CL to the CQ right away upon creation.
Add a --no-commit flag that can be specified to avoid that.
Remove the trybot status feature (--status) and the commit CL
feature (--commit).
BUG=chromium:433305
TESTED=Ran:
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /ssd/chrome/src --dry-run --ignore-checks
and verified it didn't trigger tryjobs or sent to CQ. Then I ran:
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /ssd/chrome/src --ignore-checks
and verified it was sent to CQ.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42739004
Cr-Commit-Position: refs/heads/master@{#8715}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8715 4adac7df-926f-26a2-2b94-8c16560cd09d
The AppRTCDemoTest is failing if the Android device lacks
an Internet connection (e.g. is in flight mode).
This change makes it benefit from the work done in
https://review.webrtc.org/36769004/ to work around that
limitation for loopback tests.
R=phoglund@webrtc.orgTBR=glaznev@webrtc.org
BUG=4421
TESTED=Successful run on Nexus 7 (2013) in flight mode using:
ninja -C out/Release
. build/android/envsetup.sh
adb install -r out/Release/apks/AppRTCDemo.apk
webrtc/build/android/test_runner.py instrumentation --test-apk AppRTCDemoTest --verbose --release
Review URL: https://webrtc-codereview.appspot.com/45649004
Cr-Commit-Position: refs/heads/master@{#8714}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8714 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
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.
BUG=4413
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43699004
Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
- Remove unnecessary window member from lapped_transform.
- Add comment indicated that Blocker does not take ownership of
the window passed to its constructor.
- Streamline LappedTransform constructor so members can be const.
Also use a range-based for loop in audio_processing_impl.cc for clarity.
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41229004
Cr-Commit-Position: refs/heads/master@{#8708}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8708 4adac7df-926f-26a2-2b94-8c16560cd09d
- Updated Java VideoRenderer removes setSize() from video renderer interface.
Remove no longer valid test, which requires setSize() call before any
frame can be rendered.
- test_runner.py tries to run private member of InstrumentationTestCase class.
Workaround it by renaming private loopback test method.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47549004
Cr-Commit-Position: refs/heads/master@{#8707}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8707 4adac7df-926f-26a2-2b94-8c16560cd09d
