This is a workaround - these usb headsets have one output volume
control only, labeled "Speaker". This causes the default profile
set to not control the volume at all, which is a bug. Workaround
that by creating a separate profile set.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
When a TCP socket is created the size of the send buffer (SO_SNDBUF) used is
determined by the OS, using the net.ipv4.tcp_wmem sysctl parameter. Previously
a call to setsockopt set the buffer size to a value that was too small, and
that in some cases could result in underruns and choppy playback. This
setsockopt call has now been removed so that the value determined by the OS
is used unchanged.
Note that the value used for the send buffer size is the 2nd value in
net.ipv4.tcp_wmem, e.g. if this is set to "4096 65536 8388608" the send buffer
size is set to 65536.
This adds code to specifically support legacy entries.
I kept this code in a separate commit so that it can be (relatively)
easily removed at some point in the future.
This simply exposes the formats that a device supports
via a simple protocol extension that will allow clients
to setup what a connected receiver supports format wise.
This has the advantage of allowing versioned updates in the future,
thus allowing us to be more user friendly going forward (as opposed
to just ignoring entries from old versions).
The primary motivation for this, however, is to allow variable length
storage in each entry which will be needed for upcoming work.
At present this commit will ignore any legacy entries but support
for reading and subsequently converting legacy entries will be added
shortly.
The previous logic in ade0a6f884
does not work with for input volumes.
This was discussed on the mailing list:
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2011-May/010091.html
This approach can introduce a problem when setting the volumes
for sources. What follows is Tanu Kaskinen's analysis:
[quote]
I'll quote the log:
D: protocol-native.c: Client pavucontrol changes volume of source alsa_input.pci-0000_00_1b.0.analog-stereo.
D: alsa-source.c: Requested volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -20.71 dB 1: -20.71 dB
D: alsa-source.c: Got hardware volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB
D: alsa-source.c: Calculated software volume: 0: 101% 1: 101% (accurate-enough=no)
D: alsa-source.c: in dB: 0: 0.29 dB 1: 0.29 dB
D: source.c: Volume going up to 29273 at 270475970821
D: source.c: Volume change to 29273 at 270475970821 was written 34 usec late
D: alsa-source.c: Written HW volume did not match with the request: 0: 45% 1: 45% (request) != 0: 42% 1: 42%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB (request) != 0: -22.50 dB 1: -22.50 dB
Looking at the last line, the requested volume seems to hit exactly the
right step (-21.00dB), but for some reason Alsa decides to choose
something else. I'm pretty sure that this happens because of rounding
errors. In the first phase we ask Alsa what dB value we should set, and
it returns -21.00 dB. The value is given as a long int, but we convert
that to pa_cvolume. Then when we set the volume, we convert the
pa_cvolume value back to a long integer. At this point I believe it gets
converted to -2101. This is not visible in the debug message for some
reason - the rounding algorithm must be different from what was used
with the pa_cvolume -> long conversion.
[/quote]
The commit after this contains a patch that addresses this issue.
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 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.
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
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.
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.
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.
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.
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.
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.
(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.
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.
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.
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.
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.
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.
