This will allow an embedder to use it directly.
Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)
R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.
BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
In addition to unittests a cast losing constness was corrected.
The tests added are:
1. Adjusting allowed_offset when robust validation is disabled should have no impact.
2. For noise free signals there should be no difference between robust validation or not.
3. Robust validation acts faster during startup.
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5361 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
Added is
- a member variable for turning robust validation on and off.
- API to enable/disable feature.
- API to check if enabled.
- unit tests for these APIs.
Not added is
- the actual functionality (separate CL), hence turning feature on/off has no impact currently.
- calls in AEC and AEC, where the delay estimator is used. This is also done in a separate CL when we know if it should be turned on in both components.
TESTED=trybots, module_unittest
BUG=
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5191 4adac7df-926f-26a2-2b94-8c16560cd09d
I don't believe I've witnessed this "feature" ever provide a benefit,
and have now collected some evidence of its harm when using the
extended filter mode. It can cause erroneous resets in two cases:
1. Some preprocessing noise suppression is enabled in the system (i.e.
"audio enhancements") that push the noise floor very low, possibly to
zero. If the filter is non-zero this condition can be triggered very
easily, and erroneously.
2. Non-zero energy in the filter before the peak impulse response can
cause a slight (and harmless) "pre-echo" in the error signal. This
becomes more significant as the peak is set further back in the filter.
This effect can cause needless resets during echo onsets.
In short, this isn't a great criterion for filter reset and has the
potential to cause serious harm. Ideally we would remove it entirely,
but in the interests of safety, can start with the extended mode.
BUG=1261
R=aluebs@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5159 4adac7df-926f-26a2-2b94-8c16560cd09d
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
The untrusted delay mode provides an option to reduce the "known delay"
parameter passed down to the core AEC. This is necessary to handle the
very low latencies observed with the Chromium audio backend on Mac.
Prior to this change, it was possible to pass a negative value. The AEC
produced good output in practice, but it turned out this tripped a
heretofore unnoticed assert in ProcessBlock().
This change avoids the assert, and maintains the good output across a
set of Mac recordings. Bit-exact in some cases, and in the remaining,
quickly converging to identical output.
The assert was hit on the last webrtc roll in Chromium in
content_browsertests on Mac.
Corresponds to:
https://chromereviews.googleplex.com/9960013
TBR=bjornv
TESTED=Verified locally that "content_browsertests
--gtest_filter=WebrtcBrowserTest.*"" passes.
Review URL: https://webrtc-codereview.appspot.com/2328005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4886 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-land: http://review.webrtc.org/2151007/TBR=bjornv@webrtc.org
Original change description:
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.
Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):
Machine/CPU Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.
Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.
Corresponds to these changes:
https://chromereviews.googleplex.com/9412014https://chromereviews.googleplex.com/9514013https://chromereviews.googleplex.com/9960013
BUG=454,827,1261
Review URL: https://webrtc-codereview.appspot.com/2295006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
