This reverts commit 9615def4b9.
This is part of the reversion of BlueZ 5 support so it can be added back
in a separate set of modules. This makes the code easier to maintain and
decrease PulseAudio's binary size.
This reverts commit 0e4c16e120.
This is part of the reversion of BlueZ 5 support so it can be added back
in a separate set of modules. This makes the code easier to maintain and
decrease PulseAudio's binary size.
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.
Make the PulseAudio tunnel behave the same way as a client
when it comes to figuring out how to connect to the current
PulseAudio daemon. This can be useful if you start a second
PulseAudio instance for e.g. network access.
Sometimes it would be nice to disable module-suspend-on-idle for
specific devices. For me the use case is to keep a HDMI sink running
all the time to avoid loss of audio when starting to play a stream to
the device (the HDMI receiver eats a bit from the beginning of the
stream when the device is opened). This is arguably a hacky solution
to the problem, but on the other hand, I think it's very sensible to
interpret negative timeout in the module-suspend-on-idle.timeout
property as disabling the suspending altogher. This is also how the
exit-idle-time configuration option behaves (negative value disables
automatic exiting).
I moved the property parsing from the timer restart function to the
function that creates the device_info objects, because if the timeout
is negative, we don't need to create the device_info object at all.
The old tunnel module duplicates functionality that is in libpulse,
due to implementing the native protocol, and the protocol code in
the old tunnel module tends to get broken every now and then, because
people forget to update the tunnel module protocol implementation
when changing the native protocol. module-tunnel-source-new avoids this
problem by using libpulse to communicate with the remote server.
This patch adds the ability to restore profiles if they are added after
card creation.
Adding profiles after card creation mainly happens for bluetooth cards.
Buglink: https://bugs.freedesktop.org/show_bug.cgi?id=65349
The old tunnel module duplicates functionality that is in libpulse,
due to implementing the native protocol, and the protocol code in
the old tunnel module tends to get broken every now and then, because
people forget to update the tunnel module protocol implementation
when changing the native protocol. module-tunnel-sink-new avoids this
problem by using libpulse to communicate with the remote server.
Signed-off-by: Alexander Couzens <lynxis@fe80.eu>
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>
Currently the biggest possible sink latency is 10 seconds. The total
latency of the loopback is divided evenly for the source, an
intermediate buffer and the sink, so if I want to test 10 s sink
latency, the total needs to be three times that, i.e. 30 seconds.
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.
u->sink->state is not yet updated, so the state must be read from
u->sink->thread_info.state. This makes pausing and resuming of the
smoother happen at the right time.
Thanks to Pierre Ossman for the patch.
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.
The source output and sink inputs should be corked if the corresponding
sink/source is suspended, as handled during module initialization. This
also needs to be handled during stream move, because the suspend state
of the destination sink/source might be different to the previous one.
This fixes the issue with an infinite number of "Requesting rewind due
to end of underrun" traces after a stream move.
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 replaces every occurrence of '){' with ') {'.
The ffmpeg source tree was excluded since it will disappear anyways.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-exec sed -i -e 's/){/) {/' {} \;
This patch replaces every occurrence of 'if(' with 'if ('.
The ffmpeg source tree was excluded since it will disappear anyways.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-exec sed -i -e 's/ if(/ if (/' {} \;
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 outputs are removed from the idxset before output_free() is
called. Trying to remove them again in output_free(), and asserting
that it should succeed caused crashing whenever outputs were freed.
This bug was introduced in commit
061878b5a4.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=65901
We need the mainloop lock to be taken around pa_mainloop_api_once() to
prevent an assert due to the defer event creation and setting of the
destroy callback not being performed atomically.
To save some CPU (in low latency scenarios), don't re-enable the
"writable" event after it has succeeded. It is very likely the next
write will succeed right away too.
This means that we always need to handle EAGAIN/EWOULDBLOCK as a
successful write of 0 bytes, so I also verified that all callers to
pa_iochannel_write handled this correctly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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.
This pushes all avahi-client code to a threaded mainloop from the PA
mainloop context. We need to do this because avahi-client makes blocking
D-Bus calls, and we don't want to block the mainloop for that long.
The only exception to this now that I don't see a workaround for is
during module unload time. However, this shouldn't be a huge problem
since in most cases, this will only happen at server shutdown time.
The bulk of the change is partitioning the data so that PA core objects
only (well, mostly) get accessed in the PA mainloop and Avahi calls
happen only in the Avahi threaded mainloop.
Bug: https://bugs.freedesktop.org/show_bug.cgi?id=58758
