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Svmp 2.x #1

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[email protected] and others added 30 commits May 30, 2014 09:28
Phenomenon:

When packet loss rate was fed to a codec that does not implement packet loss adaptive encoding, VoE logs an error.

Reason:

The ACM function SetPacketLossRate(int rate) return -1 unnecessarily too often. It was intended for more severe errors like
1. codec is not ready
2. input rate is out of range

BUG=webrtc:3413
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Review URL: https://webrtc-codereview.appspot.com/16599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6283 4adac7df-926f-26a2-2b94-8c16560cd09d
…e RTX wrapped headers.

This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.

BUG=1811
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/12609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
Required to run the binary on Android bots.

BUG=3423
[email protected]

Review URL: https://webrtc-codereview.appspot.com/15609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6285 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL adds a basic multi-threaded extention of the ACM unit test.
The test has three threads. One thread adds raw audio to the sender
side and encodes it. The next thread adds encoded RTP packets to the
receiver. The last thread pulls decoded audio out of the receiver.

[email protected]

Review URL: https://webrtc-codereview.appspot.com/15559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6286 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a new test; the failures are not due to a change in underlying code.

TBR=henrik.lundin
BUG=3426

Review URL: https://webrtc-codereview.appspot.com/19589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6288 4adac7df-926f-26a2-2b94-8c16560cd09d
This method is introduced to try to avoid inconsistent resetting of
AudioFrame members to default/uninitialized values.

Use it at the call points of DownConvertToCodecFormat(). Results in the
following minor functional changes:
- speech_activity_ is set to its uninitialized value. AFAICT, this
member isn't used at all in the capture path.
- timestamp_ is switched from -1 to 0. This member doesn't appear to be
used either in the capture path, but left a TODO for wu to change the
default value to better represent the uninitialized state.

Bonus: Don't copy the frame on error in RemixAndResample(). An error
indicates a logical fault (as pointed out by the asserts) that we should
not attempt to recover from.
BUG=3111
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/21519007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d
1. OnStateChange should not be fired if state is not changed.
2. RemotePeerRequestClose should be a no-op if it's already closed.

[email protected]
BUG=

Review URL: https://webrtc-codereview.appspot.com/21559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6290 4adac7df-926f-26a2-2b94-8c16560cd09d
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.

BUG=2168
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Review URL: https://webrtc-codereview.appspot.com/17589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.

[email protected]

BUG=

Review URL: https://webrtc-codereview.appspot.com/18489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
…ned certificates.

It used to compre a parent certificate's digest against the SDP fingerprint and caused connection failure.

BUG=3383
[email protected], [email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/17589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6294 4adac7df-926f-26a2-2b94-8c16560cd09d
It seems like this was a one time only and not a flaky test.

BUG=3376
TESTED=trybots
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Review URL: https://webrtc-codereview.appspot.com/15649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6298 4adac7df-926f-26a2-2b94-8c16560cd09d
Picking some low-hanging fruit. Add annotations for acm_crit_sect_ that
do not require lock changes. Also adding annotations for callbacks.

BUG=3401
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Review URL: https://webrtc-codereview.appspot.com/12579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6299 4adac7df-926f-26a2-2b94-8c16560cd09d
Last attempt reverted. Trying again in a different way.

This CL effectively reverts r6300.

BUG=3245
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Review URL: https://webrtc-codereview.appspot.com/20549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6301 4adac7df-926f-26a2-2b94-8c16560cd09d
This has been replaced with the Chromium Perf
Dashboard web application a long time ago.

BUG=
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Review URL: https://webrtc-codereview.appspot.com/21569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6303 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add parenthesis to make order of evaluation clearer.

BUG=
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Review URL: https://webrtc-codereview.appspot.com/12659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
… causes

The data_ optimization was a way to operate on the data directly
instead of copying it, applicable in the mono, non-float case. Since a
few audio_processing steps are already using floats (with more
hopefully to come), we don't end up benefiting from the optimization
anyway, so we might as well remove it.

BUG=
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/15539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6307 4adac7df-926f-26a2-2b94-8c16560cd09d
As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.

BUG=1205,3220
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Review URL: https://webrtc-codereview.appspot.com/13579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
The new format is at least as easy to read, and takes less space.

BUG=
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/16539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6311 4adac7df-926f-26a2-2b94-8c16560cd09d
Increase timeout and decrease test length. Also fixing a bug in the
test, and make sure the test aborts if fatal failure occurrs.

BUG=3426
[email protected]

Review URL: https://webrtc-codereview.appspot.com/13579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6312 4adac7df-926f-26a2-2b94-8c16560cd09d
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.

BUG=1788
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Review URL: https://webrtc-codereview.appspot.com/19599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
An example of botbreakage is http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/1807

> Re-enable AudioCodingModuleMtTest
> 
> Increase timeout and decrease test length. Also fixing a bug in the
> test, and make sure the test aborts if fatal failure occurrs.
> 
> BUG=3426
> [email protected]
> 
> Review URL: https://webrtc-codereview.appspot.com/13579005

[email protected]

Review URL: https://webrtc-codereview.appspot.com/19609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6314 4adac7df-926f-26a2-2b94-8c16560cd09d
[email protected] and others added 30 commits June 20, 2014 19:02
…a release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a terminated looper.

- Allocate real texture for camera preview.
- Add fps and camera frame duration logging.
- Get camera frame timestamp in Java code and pass it to jni code  so the frame timestamp is assigned as soon as possible. Jni code will not use these timestamps yet until timestamp ntp correction and zeroing in webrtcvideengine.cc will be addressed.

[email protected]

Review URL: https://webrtc-codereview.appspot.com/16729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6513 4adac7df-926f-26a2-2b94-8c16560cd09d
This change is based on RFC3264:

"Although the answerer MAY list the formats in their desired order of preference, it is RECOMMENDED that unless there is a specific reason, the answerer list formats in the same relative order they were present in the offer."

BUG=2868
TEST=unit tests and manually with munge-sdp test
[email protected]

Review URL: https://webrtc-codereview.appspot.com/14589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6514 4adac7df-926f-26a2-2b94-8c16560cd09d
Also cleaned up some unneeded stuff from webrtc/base/BUILD.gn

BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

[email protected]
[email protected]

Review URL: https://webrtc-codereview.appspot.com/20739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6515 4adac7df-926f-26a2-2b94-8c16560cd09d
Rolling to this new Chromium revision required us to introduce
a sanitizer_options similar to the one in Chromium's base
(see https://code.google.com/p/chromium/codesearch#chromium/src/base/base.gyp&l=977
and https://codereview.chromium.org/238123003) in order
to get the same defaults for ASan and LSan. Without it
compilation will break since LeakSanitizer (LSan) is enabled by
default in Clang r209387 that is pulled with this roll.

I setup so that we pull in the sanitizer_options.cc and
tsan_suppressions.cc files using DEPS, so we don't have to maintain
them separately for now. We can still use our own TSan suppressions.txt
file as we do today with no changes needed.

This roll also brings in http://crrev.com/276676 so we can enable
GN build for WebRTC.

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 272489:277350

which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

in a WebRTC checkout, gives the following relevant changes:
* third_party/android_tools 6fc0e1:c6e658
* third_party/libjpeg_turbo 263594:272637
* third_party/libyuv 1000:1007
* third_party/nss 271760:277057
* tools/gyp 1921:1927
* tools/swarming_client ae8085:aea506

The following also shows that Clang is upgraded from r206824 to r209387:
$ svn diff http://src.chromium.org/chrome/trunk/src/tools/clang/scripts/update.sh -r 272489:277350

BUG=3441
TEST=Trybots are not passing since after the recipe switch, SVN-based try jobs doesn't seem to support auto-detecting that a sync is needed if there's a DEPS change.
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/12729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6516 4adac7df-926f-26a2-2b94-8c16560cd09d
In the chromium_revision DEPS roll CL
https://review.webrtc.org/12729004/ (r6516) the addition
of the third_party/colorama was missed since our trybots
currently cannot handle DEPS changes in tryjob patches
properly.
Adding third_party/colorama/src fixes the Android build.

TEST=Passing local compile with GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm"
[email protected]
BUG=

Review URL: https://webrtc-codereview.appspot.com/12819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6517 4adac7df-926f-26a2-2b94-8c16560cd09d
The PeerConnectionEndToEndTest.DataChannelIdAssignment test fails
flakily like this:
[----------] 1 test from PeerConnectionEndToEndTest
[ RUN      ] PeerConnectionEndToEndTest.DataChannelIdAssignment
WARNING: no real random source present!
../../talk/app/webrtc/test/peerconnectiontestwrapper.cc:216: Failure
Value of: CheckForConnection()
  Actual: false
Expected: true
[  FAILED  ] PeerConnectionEndToEndTest.DataChannelIdAssignment (13215 ms)
[----------] 1 test from PeerConnectionEndToEndTest (13223 ms total)

[email protected]
BUG=

Review URL: https://webrtc-codereview.appspot.com/20759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6518 4adac7df-926f-26a2-2b94-8c16560cd09d
After rolling chromium_revision in r6516 it seems
TSan v2 turned on deadlock detection by default.
This caused a collection of tests to start failing.
This CL suppresses these failures awaiting further
investigation.

[email protected]
BUG=3509
TEST=Tests passing local execution on Linux using the
reproduction steps in the bug.

Review URL: https://webrtc-codereview.appspot.com/18609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6519 4adac7df-926f-26a2-2b94-8c16560cd09d
Match Chromium libvpx roll to fix Android bots.

[email protected]

Review URL: https://webrtc-codereview.appspot.com/12829005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6521 4adac7df-926f-26a2-2b94-8c16560cd09d
cd webrtc/base
svn diff -r 6466:66521 http://webrtc.googlecode.com/svn/trunk/talk/base >
6521.diff
patch -p0 -i 6521.diff

BUG=3379
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Review URL: https://webrtc-codereview.appspot.com/17769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6522 4adac7df-926f-26a2-2b94-8c16560cd09d
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.

I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.

I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.

BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/13749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
A few extra locks had to be acquired as a result of the annotation.

BUG=3401
[email protected]

Review URL: https://webrtc-codereview.appspot.com/15819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6524 4adac7df-926f-26a2-2b94-8c16560cd09d
Was causing warnings in Chromium such as:

warning C4742: 'WebRtcAec_overDriveCurve' has different alignment in 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core_sse2.c' and 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core.c': 4 and 16
warning C4744: 'WebRtcAec_overDriveCurve' has different type in 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core_sse2.c' and 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core.c': 'array (260 bytes)' and '__declspec(align(16)) array (260 bytes)'
warning C4742: 'WebRtcAec_weightCurve' has different alignment in 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core_sse2.c' and 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core.c': 4 and 16
warning C4744: 'WebRtcAec_weightCurve' has different type in 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core_sse2.c' and 'E:\src\buildbot\build\slave\fake_slave\build\src\third_party\webrtc\modules\audio_processing\aec\aec_core.c': 'array (260 bytes)' and '__declspec(align(16)) array (260 bytes)'

BUG=https://code.google.com/p/chromium/issues/detail?id=336620
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/15869004

Patch from Sebastien Marchand <[email protected]>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6525 4adac7df-926f-26a2-2b94-8c16560cd09d
A few locks had to be acquired to fully annotate the class, and a few
others had to be moved.
Removing an API method that was not used.

BUG=3401
[email protected]

Review URL: https://webrtc-codereview.appspot.com/12759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6526 4adac7df-926f-26a2-2b94-8c16560cd09d
The performance gain on a Nexus 7 reported by audioproc is ~4.7%

The output is NOT bit exact. Any difference seen is +-1.

BUG=3131
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/12779004

Patch from Scott LaVarnway <[email protected]>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6529 4adac7df-926f-26a2-2b94-8c16560cd09d
NetEq is thread-safe by virtue of it's own lock, and in r6404 the
ACMISAC class was made thread-safe. Therefore, the neteq decode lock
is no longer needed.

[email protected]

Review URL: https://webrtc-codereview.appspot.com/18599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6531 4adac7df-926f-26a2-2b94-8c16560cd09d
…ppressions which is not allowed. Renamed the talk/base ones as they are going away.

BUG=3379
[email protected]

Review URL: https://webrtc-codereview.appspot.com/13759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6532 4adac7df-926f-26a2-2b94-8c16560cd09d
…" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE].

BUG=crbug/387632
[email protected]

Review URL: https://webrtc-codereview.appspot.com/17779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6533 4adac7df-926f-26a2-2b94-8c16560cd09d
This hack was made to come around issue 845. Now that is solved, and
the test code can be cleaned up.

BUG=845
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/21709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6534 4adac7df-926f-26a2-2b94-8c16560cd09d
The performance gain on a Nexus 7 reported by audioproc is ~3.5%.

The output is bit exact.

BUG=3131
TESTED=verified performance manually, passed trybots
[email protected], [email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/12799005

Patch from Scott LaVarnway <[email protected]>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6535 4adac7df-926f-26a2-2b94-8c16560cd09d
REMB, TMMBR, TMMBN and
extended reports: RRTR, DLRR, VoIP metric.

BUG=2450
[email protected], [email protected]

Review URL: https://webrtc-codereview.appspot.com/9299005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6537 4adac7df-926f-26a2-2b94-8c16560cd09d
Conflicts:
	talk/app/webrtc/webrtcsession.cc
	talk/base/thread.h
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