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.
This patch removes all occurrences of double and triple
newlines.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name 'adrian-aec.*' -a -not \
-name reserve.c -a -not -name 'rtkit.*' \
-exec sed -i -e '/^$/{N;s/^\n$//}' {} \;
Two passes were needed to remove triple newlines.
The excluded files are mirrored files from external sources.
This patch replaces every occurrence of ')\n{' with ') {'.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name core-util.c -a -not \
-name adrian-aec.c -a -not -name g711.c \
-exec sed -i -e '/)$/{N;s/)\n{$/) {/}' {} \;
The excluded files are mirrored files from external sources.
This patch removes all tabs hidden inside the source tree and replaces
them with 4 spaces.
Command used for this:
find . -type d \( -name bluetooth \) -prune -o
-regex '\(.*\.[hc]\|.*\.cc\)' -a -not -name 'reserve*.[ch]'
-a -not -name 'gnt*.h' -a -not -name 'adrian*'
-exec sed -i -e 's/\t/ /g' {} \;
The excluded files are mirrored files from external sources containing
tabs.
The tsched_watermark is in bytes, not in usecs. Fix this by introducing
a new variable, and also use that variable in some places for optimisation.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
If there is a "Line Out" jack present, then add this path. The fallback
analog-output will be a subset of this path and removed.
I only use the "Line Out Jack" or "Line Out Front Jack" for actual jack
detection - without anything connected to the front jack, it makes little
sense to enable the port.
(Another option could perhaps be to use different paths for stereo line out
and surround line outs, but that could be a possible future improvement.)
Assume that the headphone port volume is lower than the speaker volume.
When plugging in headphones, if the path is active, while the jack is
being inserted and before it is actually detected as being plugged in,
it will still receive the signal being played (which is at a higher
volume than it will be when plugged in completely). The volume
difference manifests as a volume spike when the headphones are plugged
in, before the final volume is set.
This patch is required to prevent such a volume spike when plugging in
headphones. The problem is not fixed completely, but the spike is
shortened. To be fixed completely, we need to apply the port volume
before unmuting the new path.
Port creation is now slightly different. It is now similar to how
other objects are created (e.g. sinks/sources/cards).
This should become more useful in the future when we move more stuff to
the ports.
Functionally nothing has changed.
This means that the path names will always correspond to the
path configuration file names, so they will automatically be
unique (in the scope of one card).
Previously the path description was looked up based on the
path name only. Since there can be multiple paths that use
the same description, it had to be possible to have multiple
paths with the same name.
Having the same name with multiple paths makes identifying
the paths more complex than necessary, so the plan is to
make it impossible to have paths with the same name. This
patch prepares for that by retaining the possibility to
still have the same description with multiple paths. Instead
of the path name, the path description is looked up by using
the "path description key" if it is set (path name is still
used as a fallback lookup key).
A stationary computer usually has headphone jack(s) and line out jacks.
In some cases analog-output.conf will be a subset of
analog-output-headphones.conf, causing line outs to be unusable (because
headphones are unplugged).
This late in the cycle, this was the safest way I could think of to try
to fix this for a particular computer. In later versions of PulseAudio
we could consider making a dedicated line out path instead, and have
proper jack detection there.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
