The previous patch removed module-gconf's dependency on the userdata
pointer of the free callback, and that was the only place where the
userdata pointer of pa_free2_cb_t was used, so now there's no need for
pa_free2_cb_t in pa_hashmap_free(). Using pa_free_cb_t instead allows
removing a significant amount of repetitive code.
I was looking at a log that showed that a suspend happened (at
a strange time), but the log didn't tell me why the suspend was done.
This patch tries to make sure that that won't happen again.
This includes updating the native protocol and the client API.
A new command was added to allow setting the latency offset.
Also the card list command now shows the latency offset if there
are ports available.
Update protocol to 27.
In practice there is always at least one profile, and I
don't think there will ever be cards without profiles.
Therefore, I added assertions to pa_card_new() stating that
the card new data must always contain at least one profile.
Now a lot of code can be simplified, because it's guaranteed
that the profiles hashmap and the active_profile field are
always non-NULL.
Specifying the volume when creating a new stream is not an
equivalent act as setting the volume with a volume control
application. When creating a new stream, stream-restore
shouldn't save the volume, but when changing the volume,
then saving it is ok. For example, when I say
"paplay --volume=10000 somefile.wav", I mean that I want the
new stream to have volume 10000. I don't mean that also
future paplay invocations (without the --volume option)
should have that same volume.
This patch effectively reverts
546bcf3f2f.
As David points out, the previous commit made a couple of asserts
redundant (the XOR covers all cases that were previous tested for).
Remove these redundant commits now.
Commit 54cddc6ddf removed an assert that
looked redundant but was not. This commit reinstates it in a slightly
modified form. It is not stated as (a ^ b) instead of (!a || !b) in
order to make the condition more obvious.
For some reason, a badly behaving client was trying to set a NULL
port, which caused PulseAudio to crash. Add safeguards on two levels
just to be protected. (Also remove a redundant check.)
BugLink: https://bugs.launchpad.net/bugs/951273
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
For volume control UIs to be able to show ports in inactive profiles,
expose all ports together with the card info. This includes updating
the protocol and the client API to show the connection between ports
and for which profiles the ports are relevant.
Update protocol to 26.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This is actually implemented in module-protocol-stub as
'auth-group-enable'. An unfortunate typo because the other argument is
spelt as 'enabled', but it's better to be slightly inconsistent than to
change the public interface.
This adds a boolean module parameter to disable automatic dynamic
latency readjustments on underruns, but leaves automatic dynamic
watermark readjustments untouched.
These are not used for anything at this point, but this
makes it easy to add ad-hoc debug prints that show the
memblockq name and to convert between bytes and usecs.
This patch introduces some extra protocol information, so protocol
version is bumped. This functionality is primarily needed to solve
a long standing issue in alsa-plugins, which should ignore underruns
if and only if it is obsolete, i e, if more data has been written to
the pipe in the meantime (which will automatically end the underrun).
BugLink: http://bugs.launchpad.net/bugs/805940
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
For source and sinks, we allow both the index and name to be invalid
while introspecting (the default source/sink will be used). This
modifies the assert appropriately. Thanks to Drew Ogle
<dantealiegri@gmail.com> for reporting and providing a patch.
This piggy backs onto the previous changes for protocol 22 and
thus does not bump the version. This and the previous commits should be
seen as mostly atomic. Apologies for any bisecting issues this causes
(although I would expect these to be minimal)
We were calculating new latency based on the latency set on the old
sink/source, rather than the actual latency requested by the client.
Over a series of moves, this will lead the latency being ~halved each
time, resulting in an eventual rewind flood from a latency that cannot
be handled.
The proplist used may never be freed if an error condition was found with
CHECK_VALIDITY macro and the formats idxset was never freed regardless
of error state.
This change fixes adds a new maco CHECK_VALIDITY_GOTO() which allows
for cleanup to be done before returning.
We frequently need to free an idxset containing pa_format_infos, so
define an internal free function that can be used directly with this
(instead of defining it once-per-file).
This is the beginning of work to support compressed formats natively in
PulseAudio. This adds a pa_stream_new_extended() that takes a format
structure, sends it to the server (=> protocol extension) and has the
server negotiate with the appropropriate sink to figure out what format
it should use.
This is work in progress, and works only with PCM streams. Actual
compressed format support in some sink needs to be implemented, and
extensive testing is required.
More details on how this is supposed to work is available at:
http://pulseaudio.org/wiki/PassthroughSupport
This is pretty cosmetic change; there's no actual functionality added.
Previously the volume_writable information was available through the
pa_sink_input_is_volume_writable() function, but I find it cleaner to have a
real variable.
The sink input introspection variable name was also changed from
read_only_volume to volume_writable for consistency.
When we have a filter sink that does some processing, currently the
benefits of the flat volume feature are not really available. That's
because if you have a music player that is connected to the filter sink,
the hardware sink doesn't have any idea of the music player's stream
volume.
This problem is solved by this "volume sharing" feature. The volume
sharing feature works so that the filter sinks that want to avoid the
previously described problem declare that they don't want to have
independent volume, but they follow the master sink volume instead.
The PA_SINK_SHARE_VOLUME_WITH_MASTER sink flag is used for that
declaration. Then the volume logic is changed so that the hardware
sink calculates its real volume using also the streams connected to the
filter sink in addition to the streams that are connected directly to
the hardware sink. Basically we're trying to create an illusion that
from volume point of view all streams are connected directly to the
hardware sink.
For that illusion to work, the volumes of the filter sinks and their
virtual streams have to be managed carefully according to a set of
rules:
If a filter sink follows the hardware sink volume, then the filter sink's
* reference_volume always equals the hw sink's reference_volume
* real_volume always equals the hw sink's real_volume
* soft_volume is always 0dB (ie. no soft volume)
If a filter sink doesn't follow the hardware sink volume, then the filter
sink's
* reference_volume can be whatever (completely independent from the hw sink)
* real_volume always equals reference_volume
* soft_volume always equals real_volume (and reference_volume)
If a filter sink follows the hardware sink volume, and the hardware sink
supports flat volume, then the filter sink's virtual stream's
* volume always equals the hw sink's real_volume
* reference_ratio is calculated normally from the stream volume and the hw
sink's reference_volume
* real_ratio always equals 0dB (follows from the first point)
* soft_volume always equals volume_factor (follows from the previous point)
If a filter sink follows the hardware sink volume, and the hardware sink
doesn't support flat volume, then the filter sink's virtual stream's
* volume is always 0dB
* reference_ratio is always 0dB
* real_ratio is always 0dB
* soft_volume always equals volume_factor
If a filter sink doesn't follow the hardware sink volume, then the filter
sink's virtual stream is handled as a regular stream.
Since the volumes of the virtual streams are controlled by a set of rules,
the user is not allowed to change the virtual streams' volumes. It would
probably also make sense to forbid changing the filter sinks' volume, but
that's not strictly necessary, and currently changing a filter sink's volume
changes actually the hardware sink's volume, and from there it propagates to
all filter sinks ("funny" effects are expected when adjusting a single
channel in cases where all sinks don't have the same channel maps).
This patch is based on the work of Marc-André Lureau, who did the
initial implementation for Pulseaudio 0.9.15.
There are two known cases where read-only or non-existing sink input volume is
relevant: passthrough streams and the planned volume sharing logic.
Passthrough streams don't have volume at all, and the volume sharing logic
requires read-only sink input volume. This commit is primarily working towards
the volume sharing feature, but support for non-existing sink input volume is
also added, because it is so closely related to read-only volume.
Some unrelated refactoring in iface-stream.c creeped into this commit too (new
function: stream_to_string()).
If many small blocks are in queue, handle_seek is being called
for every one of them, sometimes causing a rewind. Delay the
call until all blocks are handled, then call handle_seek only
once.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Allow a message in the queue to perform both a seek and a post data.
For clients that do not use PA_SEEK_RELATIVE (e g gstreamer), this
cuts the message count - and sometimes even the rewinds - in half.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
