This piggy backs onto the previous changes for protocol 22 and
thus does not bump the version. This and the previous commits should be
seen as mostly atomic. Apologies for any bisecting issues this causes
(although I would expect these to be minimal)
We were using the block size in bytes instead of samples, which meant
preprocessing was broken. This fix makes a large-ish difference in the
quality of echo-cancellation with speex.
The echo suppress attenuation value was being incorrectly modified.
Fixed and added 2 arguments to change the attenuation of the residual
echo filter. Default values of the speex preprocessor will be used when
omitted.
This allows the selective enabling of speex' preprocessing algorithms
before running the echo-canceller -- for now this includes automatic
gain control, noise suppression and echo suppression. It's all off by
default for now, though at some point in the near future we might want
to enable at least denoising by default.
The denoising works pretty well, though we might want to add a way to
tweak the noise-suppression knob that libspeex provides.
The AGC option is just a stop-gap -- we need a real AGC mechanism that
tweaks the source volume rather than doing this in software.
The speex documentation mentions VAD and dereverb, but it appears that
these are not complete yet.
We don't do all this in a separate module from module-echo-cancel to
avoid the overhead of adding another virtual source. It makes more sense
to make a separate virtual source module that can be used for cases
where preprocessing is useful but AEC is not (for e.g. noise suppression
for fan noise in a recording application).
Another reason to keep this integrated with the AEC module is that the
echo suppression bits use the speex echo canceller state. This does leak
some information about the AEC implementation into module-echo-cancel,
but this is unavoidable.
This is the beginning of work to support compressed formats natively in
PulseAudio. This adds a pa_stream_new_extended() that takes a format
structure, sends it to the server (=> protocol extension) and has the
server negotiate with the appropropriate sink to figure out what format
it should use.
This is work in progress, and works only with PCM streams. Actual
compressed format support in some sink needs to be implemented, and
extensive testing is required.
More details on how this is supposed to work is available at:
http://pulseaudio.org/wiki/PassthroughSupport
(Based on Colin's review) We mark modules as being autoloaded so that
they can handle this as a special case if needed (which is required by
module-echo-cancel for now). This inverts how things were done and makes
using these modules manually less error-prone.
With automaticl filter loading by module-filter-apply, setting the
virtual sink/source to have the "phone" intended role will break routing
when you first connect a phone stream to an ALSA device and then turn on
your Bluetooth headset. This happens because module-intended-roles
doesn't move a stream if it is already on a device that provides the
required role.
This patch introduces a "manual_load" parameter that is meant to be used
when not using module-filter-apply for loading the AEC module. If this
parameter is set, the virtual devices are given the "phone" role, else
we count on module-filter-heuristics to do the right thing.
Mostly warnings about unused stuff.
Furthermore, the first hunk is a fix for the change in 177948a6.
Finally, comment in AEC_dtd was translated and the code simplified slightly.
CC module_bluetooth_device_la-module-bluetooth-device.lo
modules/bluetooth/module-bluetooth-device.c: In function ‘a2dp_process_render’:
modules/bluetooth/module-bluetooth-device.c:1335:30: warning: pointer targets in passing argument 6 of ‘sbc_encode’
differ in signedness [-Wpointer-sign]
../src/modules/bluetooth/sbc/sbc.h:92:9: note: expected ‘ssize_t *’ but argument is of type ‘size_t *’
CC module_rygel_media_server_la-module-rygel-media-server.lo
modules/module-rygel-media-server.c:383:13: warning: ‘append_property_dict_entry_object_array’ defined but not used [-Wunused-function]
CC module_echo_cancel_la-adrian-aec.lo
modules/echo-cancel/adrian-aec.h:360:15: warning: ‘AEC_getambient’ defined but not used [-Wunused-function]
modules/echo-cancel/adrian-aec.h:368:14: warning: ‘AEC_setgain’ defined but not used [-Wunused-function]
modules/echo-cancel/adrian-aec.h:374:14: warning: ‘AEC_setaes’ defined but not used [-Wunused-function]
modules/echo-cancel/adrian-aec.h:377:16: warning: ‘AEC_max_dotp_xf_xf’ declared ‘static’ but never defined [-Wunused-function]
CC module_echo_cancel_la-module-echo-cancel.lo
modules/echo-cancel/module-echo-cancel.c: In function ‘time_callback’:
modules/echo-cancel/module-echo-cancel.c:266:12: warning: variable ‘fs’ set but not used [-Wunused-but-set-variable]
CC module-virtual-sink.lo
modules/module-virtual-sink.c: In function ‘sink_input_pop_cb’:
modules/module-virtual-sink.c:206:15: warning: variable ‘current_latency’ set but not used [-Wunused-but-set-variable]
This change doesn't add any functionality in itself, but it will be useful in
the future for operating on chains of sinks or sources that are piggy-backing
on each other.
For example, the PA_PROP_DEVICE_MASTER_DEVICE property could
be handled in the core so that each virtual device doesn't have to maintain it
separately. By using the origin_sink and destination_source pointers the core
is able to see at stream creation time that the stream is created by a virtual
device, and then update that device's property list using the name of the
master device that the stream is being connected to. The same thing can be done
also when the stream is being moved from a device to another, in which case the
_MASTER_DEVICE property needs updating.
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.
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.
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.
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).
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.
