BlueZ 4 is no longer supported by BlueZ community for a long long time,
also by moving to BlueZ 5 it should make it even more clearer that
BlueZ 4 is no longer an option.
Attempt to use Acquire method if available since it directly returns
the fd in the reply or an error if that the connection could not be
created while Connect offer neither of these and depend on
NewConnection to deliver the fd.
When a new card shows up (during pulseaudio startup or hotplugged),
pulseaudio needs to pick the initial profile for the card. Unavailable
profiles shouldn't be picked, but module-alsa-card sometimes marked
unavailable profiles as available, causing bad initial profile choices.
This patch changes module-alsa-card so that it marks all profiles
unavailable whose all output ports or all input ports are unavailable.
Previously only those profiles were marked as unavailable whose all
ports were unavailable. For example, if a profile contains one sink and
one source, and the sink is unavailable and the source is available,
previously such profile was marked as available, but now it's marked as
unavailable.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102902
Currently the loopback module uses sample spec and channel map of the
sink by default. It leads to double resample if source and sink sample
specs are different and no rate/format specified in arguments. This
patch causes the source sample spec and channel map to be used by
default.
set_nonblock() will always set the file descriptor to non-blocking,
regardless of the nonblock argument.
This patch fixes the issue by passing the correct argument to the
fcntl() call. The bug had no impact because there is only one caller
of pa_make_fd_block() in poll-win32.c
This is a working implementation of a build with meson. The server,
utils, and most modules build with this, and it is possible to run from
a build tree and play/capture audio on ALSA devices.
There are a number of FIXMEs, of course, and a number of features that
need to be enabled (modules, dependencies, installation, etc.), but this
should provide everything we need to get there relatively quickly.
To use this, install meson (distro package, or mesonbuild.com) and run:
$ cd <pulseaudio src dir>
$ meson <builddir>
$ ninja -C <builddir>
The iec958 output uses device 2 and the iec958 input uses device 0. The
USB configuration in alsa doesn't set up the device numbers correctly,
which is why we need custom configuration in PulseAudio. Ideally this
would be fixed in alsa, but trying to get help for that wasn't
successful.
The volume_map variable was initialized only for PCM streams, but the
variable was passed to pa_cvolume_remap() also for non-PCM streams. The
volume remapping is never necessary for passthrough streams (PCM or
not), because no volume will be applied anyway, so let's skip the
pa_cvolume_remap() call for all passthrough streams.
jack->melem can be null if the jack disappears between probing the card
and the init_jacks() call. I don't know if jacks actually ever disappear
like that (seems unlikely), but this check is in any case needed as long
as init_jacks() has proper handling for the jack disappearing case
(rather than just an assert).
There was a crash report[1] that indicated that card_suspend_changed()
called report_jack_state() with a null melem. I don't know if that was
because the jack actually disappeared, or is there some other bug too.
[1] https://bugs.freedesktop.org/show_bug.cgi?id=104385
The commit "alsa-util: Set ALSA report_delay flag in pa_alsa_safe_delay()"
broke the build on ALSA versions below 1.1.0 because the time stamp
configuration function was introduced in 1.1.0.
This patch makes the usage of snd_pcm_status_set_audio_htstamp_config()
dependent on ALSA version.
dist_gsettingsdataconvert_DATA was set only if GSettings was enabled. If
the developer that generates the tarball doesn't have GSettings enabled,
pulseaudio.convert wouldn't get included in the tarball.
The schema file was not being added to the tarball even if GSettings was
enabled.
The current code does not call snd_pcm_status_set_audio_htstamp_config()
to configure the way timestamps are updated in ALSA. In kernel 4.14 and
above a bug in ALSA has been fixed which changes timmestamp behavior.
This leads to inconsistencies in the delay reporting because the time
stamp no longer reflects the time when the delay was updated if the
ALSA report_delay flag is not set. Therefore latencies are not calculated
correctly.
This patch uses snd_pcm_status_set_audio_htstamp_config() to set the
ALSA report_delay flag to 1 before the call to snd_pcm_status(). With
this, time stamps are updated as expected.
sco_process_render does not unref the memblock when it encounters an error.
This patch fixes the issue. It also changes the return value to 1 in the case
of EAGAIN. Because the data was already rendered and cannot be re-sent, we
have to discard the block.
Because the modified EAGAIN handling prevents the log message about EAGAIN
after POLLOUT from being printed, the log message was moved to
a2dp/sco_process_render().
The rewrite of the thread function does not change functionality much,
most of it is only cleanup, minor bug fixing and documentation work.
This patch also changes the send buffer size for a2dp sink to avoid lags
after temporary connection drops, following the proof-of-concept patch
posted by Dmitry Kalyanov.
Bug-Link: https://bugs.freedesktop.org/show_bug.cgi?id=58746
Additionally the patch changes the fixed latency for HSP playback from 125
to 25 ms. Tests showed that this produces better audio sync, which is
expected as HSP should have smaller latency than A2DP.
module-allow-passthrough has a (necessary) hack to replicate the default
sink selection and format negotiation from sink-input.c. One thing that
got missed in this replication is the possibility that the sink input is
not compatible with the default sink. When this happen, we now exit
gracefully.
Reconfiguration callback should also be set in case of avoiding resampling
option. This patch set the callback for every case because the callback
has already conditions to leave if it is not needed.
Also unnecessary codes of setting alternate sample rate to 0 are removed.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
This was originally planned to be done by paprefs when it starts, but
since the schema is now fully controlled by pulseaudio, it makes sense
to run the conversion from pulseaudio instead.
A new paprefs release is expected soon, and it will only support
GSettings. In order to have the default configuration work with the new
paprefs version, we need to enable GSettings by default.
If both module-gconf and module-gsettings are enabled when building
PulseAudio, both modules will be part of the default configuration. That
can cause trouble, because when the GConf database is migrated to
GSettings, the old configuration in GConf is not removed, so both
module-gconf and module-gsettings will try to load modules.
Generally it's not necessary to have both modules enabled even at build
time, so let's default to having only one of them enabled.
g_settings_get_child() returns a new GSettings object that needs to be
freed when it's not used any more. This patch collects all the childern
to a GPtrArray and frees them at the end of main(). They can't be freed
earlier, because that would prevent the "changed" signals from being
delivered.
The removed g_signal_connect() call didn't make sense. The callback
expects to be called when individual module groups are changed, not when
the top level object is changed. Also, module_group_callback() expects
user_data to be non-NULL, but here it was set to NULL.
According to the documentation of g_settings_list_children(), the listed
children may be removed at any time, so g_settings_get_child() may
return NULL. This is probably very unlikely to happen in practice, but
it's good to check anyway.
It is confusing if there's a thing named "module" which defines up to 10
modules to load. Calling the thing a "module group" instead should make
it easier to understand.
Originally the idea was to provide the "modules" schema with paprefs,
but since module-gsettings refers to the "modules" schema in its code,
that would make module-gsettings depend on paprefs, which is not good.
Now all schemas are provided by module-gsettings, so the paprefs
dependency is avoided. Unfortunately this means that if paprefs is
modified to load some new modules, the schema in pulseaudio needs to be
updated as well.
This also makes the module-gconf section conditional on HAVE_GCONF,
because if only gsettings support is built, the gconf section in the
configuration file would be redundant and confusing.
GConf is deprecated, and distributions are removing it. paprefs depends
on GConf, so in order to avoid paprefs getting removed as well, paprefs
has to be changed to use something else than GConf. GSettings is the
easiest alternative to migrate to, although it has the same problems
that GConf had: no support for system mode or networking.
This patch takes the non-GConf specific code from module-gconf and puts
it in stdin-util.[ch], which is then reused by module-gsettings.
module-gsettings is designed to be very similar to module-gconf.
Migration is expected to happen as follows: Distributions update
PulseAudio and paprefs at the same time, or first PulseAudio and then
paprefs. paprefs depends on module-gsettings, and module-gsettings
conflicts with module-gconf. Therefore module-gconf gets automatically
removed during the paprefs update. After the update an old PulseAudio is
likely to be running with module-gconf loaded. If the user tries to use
paprefs during this period, whatever the user does in paprefs won't have
any effect until PulseAudio is restarted (probably by a reboot or
relogin). This is not ideal, but will have to do.
When module-gsettings is loaded, it runs gsettings-data-convert
(implemented in a later patch). That will copy the settings from GConf
to GSettings. If gsettings-data-convert is not available (it's part of
GConf, so it may have already been uninstalled), then any previous
paprefs settings are lost.
