Now that we have switched to using the mixer handle only,
there is no use for sending hctl handles around.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Use the new mixer API to get callbacks, instead of using the hctl
API. Using the hctl API caused a memory leak, because alsa-lib itself
used the hctl callbacks, which we were previously overriding.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Instead of using the hctl interface, we can find controls belonging
to other iface types than "mixer". We do this by introducing a new
mixer class "SND_MIXER_ELEM_PULSEAUDIO" and create snd_mixer_elem's
for all PCM and CARD iface types (as Jacks are of the CARD type and
ELD controls are of the PCM type).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Recognize the Dock headphone jack in the same way the normal & front
headphone jacks are detected.
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Valgrind is not correctly handling ALSA TLV syscalls, which leads
to false warnings, looking like this:
"Conditional jump or move depends on uninitialised value(s)"
Unfortunately, alsa-lib itself also uses these values which valgrind
falsely believe are uninitialized, so not all warnings are removed,
but this is what we can do from PA until the valgrind bug is fixed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
In the (theoretical) case that no other elements exists but
"Line HP Swap", the presence of that element signals that there are
headphone and line-out outputs, otherwise there would be nothing to
swap.
With the new multichannel profile, we can remove this one and
handle the four channel input as a generic multichannel fallback.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The old logic assumed that if path A was a subset of path B, the
element list in B would have all elements of A in the beginning of
B's list, in the same order as A. This assumption was invalid, causing
some subset cases to not get detected. We need to search through the
full element list of B every time before we can conclude that B
doesn't have the element that we're inspecting.
In case all other profiles fail, try this fallback mapping as well.
It allows the device to specify the channel count, so it can be used
for devices that only supports being opened in multichannel mode.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
A fallback mapping or profile will only be considered for probing
if all non-fallback profiles fail.
If auto-profiles are used, a profile made up of one non-fallback
mapping and one fallback mapping will be considered a fallback profile.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Allow a mapping to relax the exact channel restriction:
exact-channels = yes | no # If no, and the exact number of channels is not supported,
# allow device to be opened with another channel count
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Remove extra-hdmi.conf, as the performance reasons behind it are invalid
Add 7.1 profiles
Add extra HDMI devices, for a total of 8
Add DTS-encoded profiles (they need dcaenc from git)
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
Surround 2.1 is one of the more common surround profiles these days,
so it's about time we support it.
The "surround21" was added to alsa-lib a few months ago, and there
hasn't yet been an alsa-lib release since, but I doubt it will change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Forcing all mute changes to go through set_mute() makes it easier to
check where the muted field is changed, and it also allows us to have
only one place where notifications for changed mute are sent.
On Haswell hardware, there are multiple HDMI outputs capable of
digital sound output. As they were identically named, KDE's control
center was unable to distinguish them, restored the wrong profile and
thus routed sound to the wrong HDMI monitor.
Also, having identically-named menu items in other mixer applications
looks like a bug.
hwparams_copy needs to be reset (as it is also reset for the third and
fourth try) before the second try.
If the reset is not done and the first try fails:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_period_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set only period size (to 1102 samples).
We have three failures and finally the fourth (only period size) succeed.
With this patch:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set period size first (to 1102 samples), buffer size second (to 4408 samples).
We only fail with the first try, the second (period followed by buffer) is
fine.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
When given an explicit device.description in card_properties, prefer
this information over other default prefixes (e.g. 'Built-in Audio')
when constructing sink/source descriptions.
For example, if I manually configure the card description to be
"FooBar", I then expect that the sinks and created by the card also
have "FooBar" in their description instead of generic "Built-in
Audio".
In some cases, "Analog Input" could show up as well as
"Headset Mic" (or "Headphone Mic"), because I forgot to add the
relevant "required-absent" lines when I added the headset mic path.
As a result, both "Analog Input" and "Headset Mic" showed up on the
Logitech USB 530 Headset.
Reported-by: Steve Magoun <steve.magoun@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
If mixer_handle is not NULL, then hctl_handle won't be NULL either.
The redundant check was confusing, because it looked like we would
leak the mixer_handle if mixer_handle is non-NULL and hctl_handle is
NULL.
The modargs are in both cases (a succesfull as well as a failed module
initialization) freed already in pa__done().
To avoid leaking modargs memory before they are assigned to u->modargs, the
code is reorganized to first allocate userdata, and then allocate the modargs.
Local variable ma is not needed anymore.
discussion here
http://lists.freedesktop.org/archives/pulseaudio-discuss/2013-December/019661.html
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Reported-by: poljar (Damir Jelić) <poljarinho@gmail.com>
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
channel count values.
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
sample rate values.
PCM Devices which have the BATCH flag set update the PCM pointer only with
period size granularity. Using timer based scheduling does not have any
advantage in this mode. For one devices which have that flag set usually update
the position pointer in software after getting the period interrupt. So
disabling the period interrupt is not possible for this kind of devices.
Furthermore writing to or reading from the buffer slice for the current period
is not possible since the position inside the buffer is not known. On the other
hand the tsched algorithm seems to get easily confused for this kind of
hardware, which results in garbled audio output. This typically means that timer
based scheduling needs to be manually disabled on systems with such devices.
Auto disabling tsched in this case allows these systems to run with the default
configuration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
These kcontrol names have started to show up lately, in
combination with surround internal speakers.
BugLink: https://bugs.launchpad.net/bugs/1236965
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
module-alsa-{sink,source}.c call pa_alsa_{sink,source}_new with
mapping set to NULL. Guard against this, like the rest of the
function does.
module-alsa-card does not use NULL, so this went unnoticed so far.
This is a cleaner solution, because it also removes paths that are
being removed because they are subsets of other paths.
Otherwise, the lingering paths could cause jack detection related
assertion failures.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=69676
Reported-and-tested-by: Kalev Lember <kalevlember@gmail.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
According to coding style, one should have one assertion per line
and not combine assertions.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
At the moment, port names combined from multiple devices are generated
based on the order that the devices are specified in config. This makes
programmatic use of thsee ports a bit painful, so let's make them be
combined in alphabetical order.
Add new PlaybackRate/CaptureRate values for UCM that can be used to
specify custom rates for devices. This value can either be set on the
verb, which makes it apply to all devices, or on the device to override
the verb setting.
This allows mappings to override some or all of the sample_spec used to
open the ALSA device. The intention, to start with, is to use this for
devices in UCM that need to be opened at a specific rate (like modem
devices). This can be extended to allow overrides in profile-sets as
well.
Since the hashmap stores a pointer to the key provided at pa_hashmap_put()
time, it make sense to allow the hashmap to be given ownership of the key and
have it free it at pa_hashmap_remove/free time.
To do this cleanly, we now provide the key and value free functions at hashmap
creation time with a pa_hashmap_new_full. With this, we do away with the free
function that was provided at remove/free time for freeing the value.
Some HD-audio codecs (at least ALC269VB and ALC283) become quite noisy on
high Mic Boost levels. So e g, if there is a "Mic Boost" and a "Capture"
control, both ranging from 0 dB to +30 dB, you get better quality if
"Mic Boost" is 0 dB and "Capture" is +30 dB, than the other way around.
By changing the order in the configuration files, this patch makes us prefer
leaving "Mic Boost" low and "Capture" high if the user selects a medium gain.
(This is based on limited experience, and there is no guarantee that there are
no sound cards that work the other way around, and therefore this patch could
potentially regress quality on those machines. Hopefully those are fewer, so
this is what we should default to.)
BugLink: https://bugs.launchpad.net/1085402
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Usually, you want to use one input or output at a time: e g,
you expect your speaker to mute when you plug in headphones.
Therefore, the headphones+speaker port should have lower priority
and both headphones and speaker.
A practical formula to do this is 1/x = 1/xa + 1/xb + .. + 1/xn.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The log message didn't match the code, so one of them was wrong. It's
entirely possible that the code is wrong, but I didn't have the
motivation to study the code enough to understand what the code is
supposed to do.
All pa_cvolume_snprint(), pa_volume_snprint(),
pa_sw_cvolume_snprint_dB() and pa_sw_volume_snprint_dB() calls have
been replaced with pa_cvolume_snprint_verbose() and
pa_volume_snprint_verbose() calls, making the log output more
informative and the code sometimes simpler.
