| Commit message (Collapse) | Author | Age | Files | Lines |
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This is required to make sure that the source output between
module-echo-cancel and ALSA can't get plugged to the virtual source or
monitor of the virtual sink that we expose. This could be triggered by
changing the profile of the underlying ALSA device.
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This forces us to get native-endian samples in the adrian module so that
we can rely on the existing endianness conversion mechanisms instead of
doing it in the module.
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The adrian module was using home-brewed endianness conversion instead of
the appropriate mactos, and speex assumed a little-endian host. This
fixes both of these.
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This uses Orc to optimise an inner loop in the core NLMS function of the
Adrian echo canceller.
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Optimises the core inner-product function, which takes the most CPU. The
SSE-optimised bits of the adrian echo canceller only if the CPU that PA
is running on actually supports SSE.
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While the sink or source is in the suspended state, disable the timer
callback because we are not doing any echo canceling then.
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Make the echo canceler drift up to 1ms now that things are more accurate.
Add 10 samples of headroom to allow for timing inaccuracies.
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Rework the code to align capture and playback samples so that we can keep more
accurate timings.
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Marks the recording and playback streams as const in the
pa_echo_canceller->run method for clarity.
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Since all algorithms will need to specify a block size (the amount of
data to be processed together), we make this a common parameter and have
the implementation set it at initialisation time.
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This adds an "aec_method" module argument to allow us to select the AEC
implementation to use.
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This adds Andre Adrian's AEC implementation from his intercom project
(http://andreadrian.de/intercom/) as an alternative to the speex echo
cancellation routines. Since the implementation was in C++ and not in
the form of a library, I have converted the code to C and made a local
copy of the implementation.
The implementation actually works on floating point data, so we can
tweak it to work with both integer and floating point samples (currently
we just use S16LE).
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Since the source and sink specification will need to be determined by
the AEC algorithm (can it handle multi-channel audio, does it work with
a fixed sample rate, etc.), we negotiate these using inout parameters at
initialisation time.
There is opportunity to make the sink-handling more elegant. Since the
sink data isn't used for playback (just processing), we could pass
through the data as-is and resample to the required spec before using in
the cancellation algorithm. This isn't too important immediately, but
would be nice to have.
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This allows us to tweak module parameters for whichever AEC module is
chosen.
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This splits out the echo-cancelling core from the PA-specific bits to
allow us to plug in other echo-cancellation engines.
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This will make splitting out the canceller parts cleaner.
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