| Commit message (Collapse) | Author | Age | Files | Lines |
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This makes sure that we handle streams moving between sinks properly. To
do this, we change the way the filter.* properties are handled a little
bit.
Firstly, this splits up the "filter.apply" property into two properties
- "filter.want" and "filter.apply". "filter.apply" acts as before - it
bypasses module-filter-heuristics and directly tells module-filter-apply
what filters are to be applied.
"filter.want" is used to tell module-filter-heuristics what filters the
client wants. The module then decides whether to actually apply the
filter or not (for now, this makes sure we don't apply echo-cancellation
even if requested on phone sinks (where it is assumed AEC is taken care
of or is not required).
Next, we also make sure that we track whether the client set
"filter.apply" or module-filter-heuristics did - and in the latter case,
we recalculate "filter.apply" and then have module-filter-apply apply
the filter if required. This introduces some evil in the form of causing
the move_finish callback to possibly trigger another move, but we
protect for this case (with a property) to be doubly sure of not causing
an infinite loop.
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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.
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Mostly warnings about unused stuff.
Furthermore, the first hunk is a fix for the change in 177948a6.
Finally, comment in AEC_dtd was translated and the code simplified slightly.
CC module_bluetooth_device_la-module-bluetooth-device.lo
modules/bluetooth/module-bluetooth-device.c: In function ‘a2dp_process_render’:
modules/bluetooth/module-bluetooth-device.c:1335:30: warning: pointer targets in passing argument 6 of ‘sbc_encode’
differ in signedness [-Wpointer-sign]
../src/modules/bluetooth/sbc/sbc.h:92:9: note: expected ‘ssize_t *’ but argument is of type ‘size_t *’
CC module_rygel_media_server_la-module-rygel-media-server.lo
modules/module-rygel-media-server.c:383:13: warning: ‘append_property_dict_entry_object_array’ defined but not used [-Wunused-function]
CC module_echo_cancel_la-adrian-aec.lo
modules/echo-cancel/adrian-aec.h:360:15: warning: ‘AEC_getambient’ defined but not used [-Wunused-function]
modules/echo-cancel/adrian-aec.h:368:14: warning: ‘AEC_setgain’ defined but not used [-Wunused-function]
modules/echo-cancel/adrian-aec.h:374:14: warning: ‘AEC_setaes’ defined but not used [-Wunused-function]
modules/echo-cancel/adrian-aec.h:377:16: warning: ‘AEC_max_dotp_xf_xf’ declared ‘static’ but never defined [-Wunused-function]
CC module_echo_cancel_la-module-echo-cancel.lo
modules/echo-cancel/module-echo-cancel.c: In function ‘time_callback’:
modules/echo-cancel/module-echo-cancel.c:266:12: warning: variable ‘fs’ set but not used [-Wunused-but-set-variable]
CC module-virtual-sink.lo
modules/module-virtual-sink.c: In function ‘sink_input_pop_cb’:
modules/module-virtual-sink.c:206:15: warning: variable ‘current_latency’ set but not used [-Wunused-but-set-variable]
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The HAVE_CLOCK_GETTIME macro protects timespec and related functions, nothing of which is used in
pa_rtclock_from_wallclock. And silently just not converting was not the proper solution anyway.
Also add an assert in pulse/mainloop.c to report the integer overflow that was triggered by the wrong
pa_rtclock_from_wallclock. Without the assert, debugging was painful.
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Use #include "header.h" if functionality of header.h is implemented
and #include <header.h> if functionality of header.h is used.
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Only whitespace changes in here
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in UIs
This value is not a technical upper limit, it's just a 'sensible'
value that is not crazy high, but also allows software amplification
above 0dB (aka 100%) for very quiet audio sources.
We recommend that a comprehensive volume control UI should allow
users to set volumes up to this limit, although of course should
deal gracefully if the user has set the volume even higher than this
without resulting in a feedback loop that effectively limits the
upper volume.
The value chosen is +11dB. This was selected somewhat subjectively
and is very similar to the current 150% that gnome-volume-control
uses (which is ~+10.57dB).
On the plus side, we now recommend that everyone allows
'Volumes up to 11' which is pretty awesome.
http://en.wikipedia.org/wiki/Up_to_eleven
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-April/006945.html
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-April/006950.html
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This completes the client-side changes to the protocol extension
introduced by commit 99ddca89cdca9b0b92ab9870764f9211e6a82e31
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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.
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Also remove some unnecessary <time.h> headers.
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And also the reverse: around some win32 specific functionality
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The check whether POSIX socket.h or WIN32 winsock2.h must be included can be
made centrally. The downside is that some functionality of e.g. arpa/inet.h is
also implemented in winsock.h, so that some files that don't use socket
functions, but do use inet.h functions, must also include pulsecore/socket.h.
(as well as arpa/inet.h)
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Instead <pulsecore/poll.h> should be included. That file includes poll.h on
platform where it is appropriate. Also remove some unnecessary <ioctl.h>
includes.
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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()).
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This prevents the smoother attached to the stream clock from being
updated while the stream is corked, which in turn ensures that once
corking is completed, pa_stream_get_time() always returns the same value
until the stream is uncorked - i.e., the clock does not advance when the
client believes that it will not.
The actual call to pa_smoother_put() happens on things like stream
suspend/unsuspend, which trigger timing updates. This changes the
smoother coefficients, which means that a call to pa_smoother_get() for
the same value of 'x' can return different values before and after a
timing update.
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This is needed to better support out of tree builds (including
distcheck) and to ensure the necessary folders are created in the
build tree on configure and also works around an intl-tools bug
(https://bugs.launchpad.net/intltool/+bug/605826)
The Makefile.am's used are minimal (and in some cases completely
blank). At present they do not include anything interesting
with the majority of the real work still done by the monolitic
src/Makefile.am
It may make sense to start splitting out src/Makefile.am into
smaller chunks but this commit makes the minimum changes to address
the issues that result from using make distcheck and other out of
tree builds.
Note: This 'breaks' the ability to type make in e.g. the src/modules
folder and have all of PA rebuilt accordingly (this is because the
static Makefiles previously present just did a "make -C ..") which
was purportedly for use in emacs. But I'm sure there will be a better
and more robust way to configure emacs to do your builds properly if
this behaviour is still desirable.
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Due to how our branching worked out, these new features will
debut in v1.0 and not v0.9.22 which has already been released
from the stable-queue branch
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As per discussions with Lennart, we will be moving to a two-component version
number scheme when the next release is made from git master branch.
This means we will be dropping the micro version component (although
for compatibility, it will remain defined as 0 in version.h).
For more information, please see the announcement here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7921
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To make concurrent use of SW and HW volume glitchles their application
needs to be synchronized. For accurate synchronization the HW volume
needs to be applied in IO thread. This patch adds infrastructure to
delay the applying of HW volume to match with SW volume timing. To
avoid synchronization problems this patch moves many of the volume and
mute related functions from main thread to IO thread. All these
changes become active only if the sync volume flag for a sink has been
set. So, for this patch to have any effect it needs to be taken into
use by sink implementor.
Signed-off-by: Jyri Sarha <jyri.sarha@nokia.com>
Reviewed-by: Tanu Kaskinen <tanu.kaskinen@digia.com>
Reviewd-by: Colin Guthrie <cguthrie@mandriva.org>
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This decrease PA_VOLUME_MAX to be less than 2^31. We want to do this in
order to simplify signed arithmetic when applying software volume
scaling (since the volume can now always be safely treated as a signed
number).
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This ensures that we always clamp the volume to PA_VOLUME_MAX. While
this currently has no effect, it will be required for making sure we
don't exceed PA_VOLUME_MAX when its value changes in the future.
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This adds a PA_VOLUME_IS_VALID() macro for checking if a given
pa_volume_t is valid. This makes changes to the volume ranges simpler
(just change PA_VOLUME_MAX, for example, without needing to modify any
other code).
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A good many of the header files are broken into a function
reference page and an overview page. These changes add
a direct link from each function reference page to their
overview page if one exists.
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stream.h, simple.h
The words drain and flush are a little ambiguous, make it explicit as
to what happens to any existing audio.
*mainloop.h
reword *_free and *_get_api for grammar
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Mostly change "Set the volume of all channels" to
"Set the volume of the first n channels" as the first is incorrect,
it doesn't set all the channels and doesn't explain what n was for.
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This commit restores the functionality originally included in 65e807
by Leszek Koltunski.
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This commit mostly converts the X11 handling to XCB. There are still
some uses of XLib to deal with the X11 session handling modules, however all
client-side code should now be free of XLib and thus this should fix Bug #799
Note that this removes the screen-based changes by Leszek Koltunski
in 65e80, however this will be restored in due course.
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Second version after Tanu's feedback
TODO:
- notify client that volume control is disabled
- change sink rate in passthrough mode if needed
- automatic detection of passthrough mode instead of hard
coded profile names
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
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The pretty name is suspposed to be understandable by non-technical
folks, and they are generally more used to the term "Subwoofer" than
"Low Frequency Emitter", so let's change the name here.
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not know anything about
All seeks/flushes that depend on the playback buffer read pointer cannot
be accounted for properly in the client since it does not know the
actual read pointer. Due to that the clients do not account for it at
all. We need do the same on the server side. And we did, but a little
bit too extreme. While we properly have not applied the changes to the
"request" counter we still do have to apply it to the "missing" counter.
This patch fixes that.
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This allows easy overriding of a clients latency setting for debugging
purposes.
http://pulseaudio.org/ticket/753
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Make suer we check the connection state before going on, so that we can
rely that s->context->pstream is properly initialized.
https://bugzilla.redhat.com/show_bug.cgi?id=539500
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Do not subtract bytes the client sends us beyond what we requested from
our missing bytes counter.
This was mostly a thinko that caused servers asking for too little data
when the client initially sent more data than requested, because that
data sent too much was accounted for twice.
This commit fixes this miscalculation.
http://bugzilla.redhat.com/show_bug.cgi?id=534130
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