Flakiness was caused by a race condition between two atomic integers shared by two threads. Fixed by counting bad packets (those not containing the expected extension) instead of the good packets.
The CL also eliminates another possible flake by introducing a test fixture which doesn't automatically start sending audio packets when constructed.
BUG=3340,3356
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6182 4adac7df-926f-26a2-2b94-8c16560cd09d
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.
Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc
Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.
BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005
BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.
We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.
This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:
[ RUN ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.
BUG=2996
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.
Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.
BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.
R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
