The filter sources should have the same max_rewind as the master source,
but these modules didn't update max_rewind when the master max_rewind
changed.
Previously max_rewind was always set to the full hw buffer size, but
the actual maximum rewind amount is limited to the part of the hw buffer
that is in use.
The rewind request that was done when lowering the sink latency had to
be moved to happen before updating max_rewind.
The practical benefit of this change: When using a filter source on a
monitor source, the filter source latency is increased by max_rewind.
Without this change the max_rewind of an alsa sink is often
unnecessarily high, which leads to unnecessarily high latency with
filter sources.
Monitor sources themselves don't suffer from the latency issue, because
they use the current sink latency instead of max_rewind for the extra
buffer that they keep to deal with rewinds.
Coverity complained about data->sink being possibly NULL when it's
dereferenced later. It was difficult for me to figure out whether that
was a false positive or not. Hopefully the comments make it a bit
easier to reason about the code in the future.
CID: 1323591
This rejigs the update_rate() logic to encompass changes to the sample
spec as a whole, as well as passthrough status. As a result,
sinks/sources provide a reconfigure() method which allows
reconfiguration as required.
The behaviour itself is currently unchanged -- alsa-sink/-source do not
actually implement anything other than rate updates for now (nor are
they ever requested to). This can be modified in the future, to allow,
for example 24-bit output when incoming media supports it, as well as
channel count changes for passthrough sinks.
Another related change is that passthrough status is now part of
sink/source reconfiguration, and we can stop doing a suspend/unsuspend
when entering/leaving passthrough state. So that part is now divided
in two -- pa_sink_reconfigure() sets the sink in passthrough mode if
required, and pa_sink_enter_passthrough() sets up everything else
(this currently means only volumes, but could disable other processing)
for passthrough mode.
The configured adjust time does not match exactly the real adjust time. Also
the adjust time varies. To improve latency estimation use an average of the
measured adjust times instead of the configured value in all calculations.
Changes:
- Mention that source outputs have volume too.
- Don't claim that most distributions have flat volumes enabled.
- Volumes use a cubic scale, not logarithmic.
- Reword the warning about using the conversion functions on hardware
volumes. The old wording gave the incorrect impression that hardware
volumes could never be converted to dB or linear scale.
Since HSP had higher priority than A2DP, the default profile when
connecting a new headset was HSP. To me it makes more sense to default
to high-quality output. We already have some automatic policies to
switch to HSP when it's needed.
I also made the A2DP source and HSP/HFP gateway profiles have lower
priority than the A2DP sink and HSP headset profiles. The A2DP source
and HSP/HFP gateway profiles should only be activated if the remote
device initiates audio streaming, so it makes sense to have lower
priority for those profiles.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=103058
This avoids the following autoconf warning:
configure.ac:89: warning: AC_COMPILE_IFELSE was called before AC_USE_SYSTEM_EXTENSIONS
../../lib/autoconf/specific.m4:368: AC_USE_SYSTEM_EXTENSIONS is expanded from...
configure.ac:89: the top level
This reverts commit ca63fbc1d8.
I applied the patch too hastily. force-speaker.conf is supposed to be
used only when the alsa mixer doesn't contain any elements that would
indicate the existence of a speaker port, but the reverted patch is a
workaround for a different problem. On the two affected EeePC machines
the Headphone element needs to be unmuted when using speakers. The
analog-output-speaker-always path happens to do that, but that's
unintentional. analog-output-speaker was changed[1] to mute the
headphone output when using the speaker port, and
analog-output-speaker-always should have been changed too, but that was
forgotten.
The kernel driver is buggy if it has a Headphone mixer element that
mutes both headphones and speakers, so this should be fixed in alsa. If
we end up having a workaround in PulseAudio for the broken driver, it
should be implemented with a new profile set and path configuration
files.
[1] https://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=22aac4e9fdb3786178f7815a0cb2150f588b1582
This adds a port, card and profile to RAOP sinks to make it
possible to change the latency at runtime (and have it persist)
using pavucontrol or pactl set-port-latency-offset.
Also move the IP:port part of the sink name to the port name.
Correct spelling of 'through' in a comment helps to fix a warning :)
also drop some unrelated comments
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
the comment /* Fall through. */ fixes the warning, but gcc-7 seems to be more
picky about the position in the code
pulsecore/sink-input.c: In function ‘pa_sink_input_update_proplist’:
pulsecore/sink-input.c:1531:13: warning: this statement may fall through [-Wimplicit-fallthrough=]
for (state = NULL; (key = pa_proplist_iterate(i->proplist, &state));) {
^~~
pulsecore/sink-input.c:1539:9: note: here
case PA_UPDATE_REPLACE: {
^~~~
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
EAGAIN is used allover the code rather than EWOULDBLOCK
POSIX allows EAGAIN and EWOULDBLOCK to have the same value (and in fact it is)
don't check for EWOULDBLOCK
modules/raop/raop-client.c: In function ‘send_udp_audio_packet’:
modules/raop/raop-client.c:473:41: warning: logical ‘or’ of equal expressions [-Wlogical-op]
if (written < 0 && (errno == EAGAIN || errno == EWOULDBLOCK)) {
^~
modules/raop/raop-client.c: In function ‘resend_udp_audio_packets’:
modules/raop/raop-client.c:528:45: warning: logical ‘or’ of equal expressions [-Wlogical-op]
if (written < 0 && (errno == EAGAIN || errno == EWOULDBLOCK)) {
^~
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
Use pa_assert_se() to check return value (pro forma) like everywhere else
Coverity ID: #154313
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
make install was failing on Windows because there cp is
used to replace ln -s commands but cp doesn't resolve its
first argument as relative to the second one.
This patch changes the install-bashcompletion-aliases rules
to chdir to the target dir so that cp works correctly. This
is the solution recomended in the automake documentation.
The macro LADSPA_PATH was defined as a list of directories quoted but
without taking into account that the directory names, specially on
Windows, can contain backslashes that need escaping.
This patch removes the quoted from the macro and uses the C preprocessor
to quote it properly using a helper macro.
Under MinGW, LC_MESSAGES is defined in libint.h which is not
included when pulseaudio is configured with nls disabled.
LC_MESSAGES is referenced when setting PA_PROP_APPLICATION_LANGUAGE.
This patch just disables setting that property when ENABLE_NLS
is not defined.
It seems that the intention was to create create write_thread_event on
thread_mainloop instead of main_mainloop (the first parameter of
io_new() is thread_mainloop, io_free() is called on thread_mainloop
etc.).
As long as both mainloops are implemented with pa_mainloop, this bug has
no effect on behaviour, because the io_new() implementation is the same.
And indeed, with the current code base both mainloops are always
pa_mainloops. However, when the tunnel-new modules switches to pa_rtpoll
as the pa_mainloop_api provider, this bug would cause problems.
Pulseaudio tries to pick the best profile (on startup or
hotplugged), the best profile is the profile with the highest
priority which isn't unavailable.
Due to the facts that iec958 ports available status always (?)
is unknown, and that it is generally more likely that a user use
hdmi than iec958, lets prioritze hdmi over iec958.
This patch shift the analog-* mappings +5 and hdmi-* mappings +5.
Use predefined values depending on the server, and make it configurable.
AirPlay is supposed to have 2s of latency. With my hardware, this is
more 2.352 seconds after numerous tests.
Switch from pausing/resuming the smoother to resetting it because the
smoother got stuck returning the same value after an idle/running cycle,
making latency calculation wrong.
This breaks a lot of headsets, so disabling by default. Can be
re-enabled in configuration for specific hardware where it is deemed
necessary.
Also added some debug logging to be able to examine what MTU size is
reported by the device.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102660
Some sound cards don't have any alsa-lib configuration, but they used to
work well enough up to PulseAudio 10. PulseAudio 11 stopped using "hw:0"
for the analog-stereo mapping, and instead defined it as a fallback
mapping without any mixer handling. As a result, switching between
headphones and speakers stopped working without changing the mixer
settings manually at least on Toshiba Chromebook 2. This patch adds the
mixer handling back to the fallback mapping.
I also renamed "unknown-stereo" to "stereo-fallback", because I like
that name more.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102560
We're supposed to prioritize USB sound cards over PCI sound cards, but
the priority bonus for the "internal" form factor prevents this from
happening. Not all (if any) USB sound cards have the form factor
property set, whereas at least on my laptop the on-board sound card has
the form factor set to "internal".
The order of the pa_sink_input_put() and pa_sink_put() calls in filter
modules was swapped in commit edc465da77 ("virtual sources and sinks:
Don't double attach a sink input or source output on filter load").
If flat volumes and volume sharing is enabled, the pa_sink_input_put()
call will update volumes of the whole tree of virtual sinks that are
connected to the root sink. The recursive updating procedure tried to
also update the volume of the new sink for which pa_sink_put() had not
yet been called, causing an assertion failure.
This patch tries to make sure that the volume of not-yet-linked sinks
is never changed. pa_sink_put() will set the sink volume correctly, so
it's fine to skip the not-yet-linked sinks during pa_sink_input_put().
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102549
The autospawn mechanism already had a root-autospawn protection mechanism. When
using systemd that was lost. Systemd 234 has a mechanism to conditionalize unit starting
on the running user, so lets do that to protect against root autospawning.
