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These were supposed to be removed already in 13849f153, but
at that time I missed the ifdefs in
module-bluetooth-discover.c.
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The speex_preprocess_ctl() function takes a spx_int32_t, but we've been
passing a pa_bool_t, which could potentially crash.
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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.
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By using module indexes rather than module pointers we avoid this posibility.
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This makes process_render_null consistent with render_memblock and
avoids introducing slight inaccuracies in early latency estimates.
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The smoother is paused on initialization and resumed when the sink
state is set to running. Otherwise, early latency estimates are
too low since there is some delay between module initialization and
entering the running state.
After the smoother is initially resumed, it is paused when the sink
state is not running. The previous behavior was to pause only when
the sink enters suspended state, however, this would lead to large
errors in latency estimates after the sink has been idle for some
time.
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The smoother was being initialized with offset zero, which caused
the sink latency to be unconditionally reported as zero.
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Add a variable to track whether the actual volume is set or not.
Suppose this:
min volume: -126 max volume: 0
then when user wants to set some constant volume to -10, it would fail.
While the alsa values are typically positive, some values are "funky"
and have negative values. It is desirable to fix this at the alsa
level so that the numbers are positive, but it's not technically
invalid, and thus we have to support it.
Discussed here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/9832
and
http://thread.gmane.org/gmane.linux.alsa.devel/85459
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If module initialisation fails, the speex done() function might try to
free a value that's not been allocated yet. Adding protection for this
condition.
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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.
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This is how it is expected to be run.
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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.
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This was making it impossible to run in debug mode with save_aec=1
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The padding was to be 16 bytes, not 16 elements.
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Thanks to Tanu Kaskinen for pointing out the missing return.
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In a setup with one or more filter sinks or sources there is always at
least one stream existing. In such a situation normal mempool
vacuuming never happens. This patch causes suspend-on-idle module to
vacuum memory when ever it notices that all sinks and sources are
suspended. The behavior can be enabled with a module parameter.
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This updates the tunnel module for protocol version >= 19.
module-tunnel-sink does not proxy server-side passthrough support (yet).
This would require a few more changes, namely keeping track of what
formats are available and if any other sink inputs are connected on the
server-side.
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Since we currently have two mechanisms to signal a passthrough
connection (non-PCM format or PA_SINK_INPUT_PASSTHROUGH flag), we move
all the related checks into functions and use those everywhere.
This makes things more consistent, and should we decide to get rid of
the flag, we only need to change pa_sink_input_*_is_passthrough()
accordingly.
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When a passthrough sink-input is added, we need to reconfigure the
sink's sample rate since no resampling occurs. We revert to the original
rate when the passthrough sink-input is removed.
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These aren't used any more - we handle passthrough mode in the iec958*
profiles now.
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This removes the passthrough flag from sinks since we will drop
exclusively passthrough sinks in favour of providing a list of formats
supported by each sink. We can still determine whether a sink is in
passthrough mode by checking if any non-PCM streams are attached to it.
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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
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(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.
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This was just introduced for debugging and should not have been in the
final commit. Won't make a difference at the moment since this function
is used as a pointer, but removing this in case we change this in the
future.
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PA_ALIGNED can't always guarantee that the alignment we want (the GCC
man page suggests that the linker might not be able to meet the
alignment requirements we desire). Instead, we now allocate some extra
memory and guaratee that the alignment we require is met.
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(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.
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This also includes a compatibility module that will ultimately be removed in
a future release.
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In case the io thread is starting there is no point on closing SCO just
to resume it latter when the source/sink changes to opened state.
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pa_dbus_append_basic_array_variant_dict_entry should take the direct
pointer to the array.
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When using transport configured via Media API sample spec needs to be
updated since codec configuration may affect it when e.g. headset
configure a different frequency or number of channels from default.
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If u->connections isn't empty when module-dbus-protocol is
unloaded, then connection_free() is called for the
remaining connections when the idxset is freed.
connection_free() tries to remove the connection from the
idxset, but that fails, because the item has already been
removed from the idxset in this scenario.
The problem is solved by not trying to remove the connection
from the idxset in connection_free(). Instead, whoever wants
to delete connections, has to remove the connection from the
idxset in addition to calling connection_free().
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This patch adds a new update mode specifier that can be optionally
given in match rules after the regexp. Property list updates triggered
by the rule will honour the given mode. The two allowed modes are 'merge'
and 'replace', corresponding to PA_UPDATE_MERGE and PA_UPDATE_REPLACE
respectively. If omitted, the mode defaults to PA_UPDATE_MERGE, ie. to
the original behavior.
For example, to force 'media.role' to be overwritten with 'bar' for
streams matching foo you can use an entry like this:
foo replace "bar"
This will really overwrite media.role to bar even if it has already been
set to something else by the application.
Thanks to Krisztian Litkey for the original patch and the description
above. In addition to implementing the new feature, this patch fixes
a number of bugs in the parsing code.
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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.
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This makes the core code in the filter-* modules generic enough to be
used on sources or sinks. We need special handling for modules that
introduce more than one sink (for now echo-cancel only).
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This makes sure that we handle streams moving between sinks properly. To
do this, we change the way the filter.* properties are handled a little
bit.
Firstly, this splits up the "filter.apply" property into two properties
- "filter.want" and "filter.apply". "filter.apply" acts as before - it
bypasses module-filter-heuristics and directly tells module-filter-apply
what filters are to be applied.
"filter.want" is used to tell module-filter-heuristics what filters the
client wants. The module then decides whether to actually apply the
filter or not (for now, this makes sure we don't apply echo-cancellation
even if requested on phone sinks (where it is assumed AEC is taken care
of or is not required).
Next, we also make sure that we track whether the client set
"filter.apply" or module-filter-heuristics did - and in the latter case,
we recalculate "filter.apply" and then have module-filter-apply apply
the filter if required. This introduces some evil in the form of causing
the move_finish callback to possibly trigger another move, but we
protect for this case (with a property) to be doubly sure of not causing
an infinite loop.
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This makes sure that we don't apply AEC on sinks that are already
connected to a "phone" device, the assumptiong being that anything
marked as such either doesn't have need it, or handles it itself.
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This makes gdb's "info threads" better understandable
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This fixes a long standing race condition when tearing down streams on
Mac OS X.
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This value is passed on to the instances of module-coreaudio-device that
are loaded upon device detection. The value is purely optional, as the
device module will fall back to to its default if it's not given.
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Adds an autoclean option (defaults to TRUE) that controls whether
module-filter-apply cleans up unused modules or not. This is useful in
cases where you know that a filter will be used often and thus can avoid
overhead from repeated module load/unload.
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