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
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
In addition an implicit cast from int32_t to int16_t was removed, which was a bug.
BUG=3348,3353
TESTED=Locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41169004
Cr-Commit-Position: refs/heads/master@{#8665}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8665 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/47449004
Cr-Commit-Position: refs/heads/master@{#8664}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8664 4adac7df-926f-26a2-2b94-8c16560cd09d
Prevent symbols defined in assembly sources from being exported in
libraries which include them by marking them hidden, as they are
implementation details.
BUG=webrtc:4183
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36759004
Patch from Richard Coles <torne@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8658}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8658 4adac7df-926f-26a2-2b94-8c16560cd09d
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.
Not inlining virtual functions with simple bodies such as
{ return false; }
strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.
BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47429004
Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 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
In addition an implicit cast from int32_t to int16_t was removed, which was a bug.
BUG=3348, 3353
TESTED=Locally on Mac and trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41179004
Cr-Commit-Position: refs/heads/master@{#8653}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8653 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
The problem we were running into on the Mac 10.9 debug bot in Chrome turned out to be good ol'fashion memory corruption. Part of webrtc was being compiled with _DEBUG, another half without it. This caused the definition of some symbols to be out of sync (notably pthread_mutex_t) and would cause code built from common.gypi, to overwrite memory allocated via common types from base/base.gypi derived code. Fun stuff to track down. This was a problem in particular with base/criticalsection.h since it's inlined into multiple object files but will have different definitions of what a mutex is.
TBR=pbos,kjellander
BUG=
Review URL: https://webrtc-codereview.appspot.com/43659004
Cr-Commit-Position: refs/heads/master@{#8646}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8646 4adac7df-926f-26a2-2b94-8c16560cd09d
> Speculative revert of 8631 "Remove lock from Bitrate() and FrameRate() in Video..."
>
> We ran into the alignment problem on Mac 10.9 debug again. This is the only CL I see in the range that adds an rtc::CriticalSection, so I'm trying out reverting it before attempting another roll.
>
> > Remove lock from Bitrate() and FrameRate() in VideoSender.
> > These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder. It's therefore safe to not require a lock to access _encoder on this thread.
> >
> > I'm making access to the rate variables from VCMGenericEncoder, thread safe, by using a lock that's not associated with the encoder. There should be little to no contention there. While modifying VCMGenericEncoder, I noticed that a couple of member variables weren't needed, so I removed them.
> >
> > The reason for this change is that getStats is currently contending with the encoder when Bitrate() is called. On my machine, this means that getStats can take about 25-30ms instead of ~1ms.
> >
> > Also adding some documentation for other methods and a suggestion for how we could avoid contention between the encoder and the network thread.
> >
> > BUG=2822
> > R=mflodman@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/43479004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/45529004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46519004
Cr-Commit-Position: refs/heads/master@{#8645}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8645 4adac7df-926f-26a2-2b94-8c16560cd09d
