Removing the bit that automatically loads module-echo-cancel for phone
streams. Clients need to specifically opt in for this now with
filter.want, until we are reasonably certain this won't break other apps
(Skype for one, possibly others).
The order of freeing the hashmaps is important here, because otherwise a string used as key is freed before the hashmap
is freed.
Valgrind reports this as:
Invalid read of size 1
at 0x4107042: pa_idxset_string_hash_func (idxset.c:67)
by 0x4106026: remove_entry (hashmap.c:93)
by 0x41061BF: pa_hashmap_free (hashmap.c:110)
by 0x71DD143: pa_dbusiface_core_free (iface-core.c:2105)
by 0x71F2169: module_dbus_protocol_LTX_pa__done (module-dbus-protocol.c:595)
by 0x406DC51: pa_module_free (module.c:162)
by 0x406E01D: pa_module_unload_all (module.c:210)
by 0x4068842: core_free (core.c:169)
by 0x406FD5D: pa_object_unref (object.c:64)
by 0x805224D: pa_core_unref (core.h:184)
by 0x805560B: main (main.c:1159)
Address 0x4d099c0 is 0 bytes inside a block of size 100 free'd
at 0x4025BF0: free (vg_replace_malloc.c:366)
by 0x40F128C: pa_xfree (xmalloc.c:131)
by 0x71E4CEB: pa_dbusiface_device_free (iface-device.c:1293)
by 0x71DCD7E: free_device_cb (iface-core.c:2062)
by 0x41061D7: pa_hashmap_free (hashmap.c:113)
by 0x71DD125: pa_dbusiface_core_free (iface-core.c:2104)
by 0x71F2169: module_dbus_protocol_LTX_pa__done (module-dbus-protocol.c:595)
by 0x406DC51: pa_module_free (module.c:162)
by 0x406E01D: pa_module_unload_all (module.c:210)
by 0x4068842: core_free (core.c:169)
by 0x406FD5D: pa_object_unref (object.c:64)
by 0x805224D: pa_core_unref (core.h:184)
The commit 7ebc5033 resulted in segfaults, because format->plist was not allocated.
The solution is not to allocate pa_format_info on the stack, but to properly use pa_format_info_new().
Also a typo regarding pa_tagstruct_putu8 is corrected.
This makes the pa_bool_t members of userdata not be a single bit field
since pa_bool_t can be an int, potentially causing signedness issues in
comparisons.
This implements the sink get_formats() and set_formats() API in
alsa-sink. Modules can use this to allow users to specify what formats
their receivers support.
Note in protocol-dbus.c specifically, method_signatures needs to be freed
before method_handlers, because otherwise h->method_name is freed while it is
still in use as a key in the method_signatures hashmap.
revents marked as POLLOUT|POLLERR|POLLWRNORM in "underrun" case that will
trigger unexpected log "ALSA woke us up to write new data to the device, but
there was acturally nothing to write...".
This patch avoids this log message.
* Fix extension API function export list.
* Ensure we trigger a subscription event when things change.
* Send the index with our subscription events.
* Clear out any existing formats when saving.
* Call the correct extension command for subscriptions.
As reported in http://kpaste.net/04f1f3f
it is possible to call enumeration_is_subset with null pointers.
Handle that case instead of crashing. (It is also possible that
Tanuk's pending element_is_subset patch solves the issue, but this
nevertheless gives some extra security.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
We want to set the volume callbacks only if volume sharing
is not used. When volume sharing is used, we don't want to
mess with the stream volumes.
This was broken in 6c6b50
Now that subset mixer paths are removed, this workaround is no longer needed.
This effectively reverts 1c38b5d478 but due
to me forgetting to add files and adding a couple extra workarounds after,
it's easier to just do this manually rather than run git-revert.
In order to try and avoid 'spamming' the user with port choices,
attempt to detect and remove any pointless paths in a path set. That is
any paths which are subsets of other paths.
This should solve a problem case with some USB Headsets which result in
two paths both involving the 'Speaker' element. When no 'Master' element
exists (which is quite common on head/handsets), then the first path
(analog-output) will contain the 'Speaker' in a way that completely fits
with in the use of the 'Speaker' element in the other path
(analog-output-speaker).
Unification is really just a 'lowest common denominator' system. If any
paths do not support volume, mute or decibels, then mark them all as not
having them.
This was originally done this way because the flags set on sinks that
dictate if it supports h/w volume, mute etc. could not be changed after
the sink was created.
The fact that these flags could not change has now been change in the
previous commits, and thus there is now no need to use this 'lowest
common denominator' approach as we can fully support the various
different combinations, even if they change after initial creation
of the sinks/source.
Some sink flags are really just a product of what callbacks
are set on the device. We still enforce a degree of sanity
that the flags match the callbacks set, but we also set the
flags automatically in our callback setter functions to
help ensure that a) people use them and b) flags & callbacks
are kept in sync.
This is not currently useful but future commits will make further
changes concerning automatic setting of flags and event delivery
that makes this structure necessary.
This picks sane defaults for the sample spec used (32 kHz, mono) and
preprocessing (on by default). This should make it unncessary to provide
additional parameters in the default desktop case.
The main exception would be decreasing the sample rate for hardware with
limited processing power (can bring it down to 16 or 8 kHz).
This is a workaround - these usb headsets have one output volume
control only, labeled "Speaker". This causes the default profile
set to not control the volume at all, which is a bug. Workaround
that by creating a separate profile set.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
When a TCP socket is created the size of the send buffer (SO_SNDBUF) used is
determined by the OS, using the net.ipv4.tcp_wmem sysctl parameter. Previously
a call to setsockopt set the buffer size to a value that was too small, and
that in some cases could result in underruns and choppy playback. This
setsockopt call has now been removed so that the value determined by the OS
is used unchanged.
Note that the value used for the send buffer size is the 2nd value in
net.ipv4.tcp_wmem, e.g. if this is set to "4096 65536 8388608" the send buffer
size is set to 65536.
This adds code to specifically support legacy entries.
I kept this code in a separate commit so that it can be (relatively)
easily removed at some point in the future.
This simply exposes the formats that a device supports
via a simple protocol extension that will allow clients
to setup what a connected receiver supports format wise.
This has the advantage of allowing versioned updates in the future,
thus allowing us to be more user friendly going forward (as opposed
to just ignoring entries from old versions).
The primary motivation for this, however, is to allow variable length
storage in each entry which will be needed for upcoming work.
At present this commit will ignore any legacy entries but support
for reading and subsequently converting legacy entries will be added
shortly.
The previous logic in ade0a6f884
does not work with for input volumes.
This was discussed on the mailing list:
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2011-May/010091.html
This approach can introduce a problem when setting the volumes
for sources. What follows is Tanu Kaskinen's analysis:
[quote]
I'll quote the log:
D: protocol-native.c: Client pavucontrol changes volume of source alsa_input.pci-0000_00_1b.0.analog-stereo.
D: alsa-source.c: Requested volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -20.71 dB 1: -20.71 dB
D: alsa-source.c: Got hardware volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB
D: alsa-source.c: Calculated software volume: 0: 101% 1: 101% (accurate-enough=no)
D: alsa-source.c: in dB: 0: 0.29 dB 1: 0.29 dB
D: source.c: Volume going up to 29273 at 270475970821
D: source.c: Volume change to 29273 at 270475970821 was written 34 usec late
D: alsa-source.c: Written HW volume did not match with the request: 0: 45% 1: 45% (request) != 0: 42% 1: 42%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB (request) != 0: -22.50 dB 1: -22.50 dB
Looking at the last line, the requested volume seems to hit exactly the
right step (-21.00dB), but for some reason Alsa decides to choose
something else. I'm pretty sure that this happens because of rounding
errors. In the first phase we ask Alsa what dB value we should set, and
it returns -21.00 dB. The value is given as a long int, but we convert
that to pa_cvolume. Then when we set the volume, we convert the
pa_cvolume value back to a long integer. At this point I believe it gets
converted to -2101. This is not visible in the debug message for some
reason - the rounding algorithm must be different from what was used
with the pa_cvolume -> long conversion.
[/quote]
The commit after this contains a patch that addresses this issue.
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)
