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
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.
Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.
BUG=4374
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42699004
Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports. As is, that code is intertwined, so we populate the stats on both signaling and worker thread.
I'm also adding some documentation and TODOs for further improvements.
BUG=2822
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47459004
Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
This change is just to allow rolling into Chromium, update Chromium and then commit the actual change in WebRTC that requires the interface change. It allows using a StatsReport::Id object as a pointer (foo->Bar()), since in an upcoming change, Id objects will be pointers.
R=magjed@webrtc.org
BUG=2822
Review URL: https://webrtc-codereview.appspot.com/43689004
Cr-Commit-Position: refs/heads/master@{#8697}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8697 4adac7df-926f-26a2-2b94-8c16560cd09d
This reverts r8683 and is a reland of r8682.
Reason for revert: The thread checker in Chromium that crashed has been fixed now.
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/40319004
Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
* Add optional --chromium-checkout flag to point out the Chromium checkout to
work in (automatic fallback to current working dir if that's a Chromium checkout).
* Introduce logging and --verbose flag.
* Make functions private and adopt Python style guide.
* Made --dry-run perform the presubmit checks and other stuff for --commit, except the actual commit.
* Add EDITOR=true environment to 'git cl upload' to avoid launching an interactive editor.
* Removed interactive invocation of 'git cl upload' and 'git cl land'
BUG=chromium:433305
TESTED=Muliple local runs using:
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /path/to/chrome/src --dry-run --verbose
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /path/to/chrome/src --dry-run --abort
tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /path/to/chrome/src --dry-run --status
cd /path/to/chrome/src
/path/to/webrtc/src/tools/autoroller/roll_webrtc_in_chromium.py --dry-run --verbose
R=phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46559004
Cr-Commit-Position: refs/heads/master@{#8694}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8694 4adac7df-926f-26a2-2b94-8c16560cd09d
Landing this on behalf of malmnas@.
The semantics is as follows:
* if the output filename is empty, then don't log to file
* if the input filename is empty, then don't stream any audio from file
This is useful for long running tests with multiple participants.
With logging turned on, having 10 bots running for 2 hours results in
more then 7 GB of data.
BUG=None
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41219004
Cr-Commit-Position: refs/heads/master@{#8691}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8691 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the need for scoped_ptr<I420VideoFrame> in VieCapturer.
This adds the method I420VideoFrame::Reset and replace the use of scoped_ptr in ViECapturer.
Also, a unittest is added to check that ViECapturer does not retain a frame after it has been delivered.
BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43669004
Cr-Commit-Position: refs/heads/master@{#8678}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8678 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding a pristine copy (only changed license and ownerless
TODOs) of the script we've used to roll the WebRTC revision
in Chromium DEPS.
This script assumes being located inside a Chromium checkout,
so a follow-up CL will be created for review of the required
changes for that before it can be used.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47499004
Cr-Commit-Position: refs/heads/master@{#8675}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8675 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
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,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)
- minor cleanups like remove of unnecessary parentheses and style changes
BUG=3348,3353
TESTED=Locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41149004
Cr-Commit-Position: refs/heads/master@{#8666}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8666 4adac7df-926f-26a2-2b94-8c16560cd09d
