Tanu Kaskinen [Fri, 11 Mar 2011 11:41:28 +0000 (13:41 +0200)]
alsa-mixer: When figuring out the max_dB of a path, use only channels that are used by the path elements.
Without this, p->max_dB could never be less than 0 dB, because the loop at the
end of pa_alsa_path_probe() would reset p->max_dB to 0 as soon as the loop
encountered a channel that wasn't touched by any element.
There was a similar issue for p->min_dB too (it could never be more than 0 dB),
which is also fixed by this patch.
Tanu Kaskinen [Fri, 11 Mar 2011 11:40:51 +0000 (13:40 +0200)]
alsa-mixer: Implement support for setting element specific upper limits for volume.
This feature is mainly useful in embedded systems that have built-in speakers.
In such situations the full audio path is known beforehand, so it's possible to
know what is the maximum sensible volume, and any higher volume can be
disabled.
The volume limit is set in path configuration files in the [Element] section,
using option "volume-limit". The value is the desired maximum volume step of
the volume element.
Arun Raghavan [Mon, 25 Oct 2010 16:59:08 +0000 (17:59 +0100)]
volume: Add Orc-based optimised volume scaling
This adds volume scaling for 1- and 2-channel software volume scaling
using Orc. While testing the MMX and SSE backends on a Core2, I see an
~2x performance benefit over the hand-rolled MMX and SSE code. Since I
haven't been able to test on other architectures, the Orc code is only
used when MMX/SSE* is present. This can be changed in the future after
testing on AMD and ARM machines.
Arun Raghavan [Wed, 27 Oct 2010 10:30:14 +0000 (11:30 +0100)]
volume: Fix sample array size for tests
Somewhere in the history of the MMX tests, the number of channels was
changed from 1 to 2, but the number of samples was not increased to make
it even (multiple of the frame size).
It seems git managed to mess up a git-am with a patch from
David which moved where this function was called element_probe
to within itself (recursive which could normally lead to an
infinite loop, but as it was now never called from anywhere else,
this didn't happen).
Thank you to Maarten for spotting and following up the issue.
Colin Guthrie [Thu, 3 Mar 2011 12:04:31 +0000 (12:04 +0000)]
volume: Add a PA_VOLUME_UI_MAX define for the recommended max volume to show 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
Colin Guthrie [Tue, 1 Mar 2011 18:15:28 +0000 (18:15 +0000)]
x11: Make pax11publish -r remove PULSE_SESSION_ID
This is not set by pax11publish, but module-x11-publish does so this tool
should tidy that up. It is only removed when passing -r and is
ignored when actually setting up it's own properties from the conf
files/guesswork.
Tanu Kaskinen [Thu, 24 Feb 2011 14:16:43 +0000 (16:16 +0200)]
virtual-sink: Fix a crash when moving the sink to a new master right after setup.
If the virtual sink is moved to a new master right after it has been created,
then the virtual sink input's memblockq can be rewound to a negative read
index. The data written prior to the move starts from index zero, so after the
rewind there's a bit of silence. If the memblockq doesn't have a silence
memchunk set, then pa_memblockq_peek() will return zero in such case, and the
returned memchunk's memblock pointer will be NULL.
That scenario wasn't taken into account in the implementation of
sink_input_pop_cb. Setting a silence memchunk for the memblockq solves this
problem, because pa_memblock_peek() will now return a valid memblock if the
read index happens to point to a hole in the memblockq.
I believe this isn't the best possible solution, though. It doesn't really make
sense to rewind the sink input's memblockq beyond index 0 in the first place,
because now when the stream starts to play to the new master sink, there's some
unnecessary silence before the actual data starts. This is a small problem,
though, and I don't grok the rewinding system well enough to know how to fix
this issue properly.
I went through all files that call pa_memblockq_peek() to see if there are more
similar bugs. play-memblockq.c was the only one that looked to me like it might
be broken in the same way. I didn't try reproducing the bug with
play-memblockq.c, though, so I just added a FIXME comment there.
Tanu Kaskinen [Thu, 24 Feb 2011 14:16:38 +0000 (16:16 +0200)]
Implement the "volume sharing" feature.
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.
David Henningsson [Mon, 20 Dec 2010 11:29:27 +0000 (12:29 +0100)]
alsa-mixer: add required-any and required-* for enum options
Now you can add required-any to elements in a path and the path
will be valid as long as at least one of the elements are present.
Also you can have required, required-any and required-absent in
element options, causing a path to be unsupported if an option is
(not) present (simplified example: to skip line in path if
"Capture source" doesn't have a "Line In" option).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
David Henningsson [Mon, 20 Dec 2010 10:13:37 +0000 (11:13 +0100)]
alsa-mixer: Add a few well-known descriptions
Add front mic, rear mic, and docking line-in. These are likely to be
present on modern hda chips, for reference see
linux-2.6/sound/pci/hda/hda_codec.c:hda_get_input_pin_label
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tanu Kaskinen [Fri, 25 Feb 2011 14:27:27 +0000 (16:27 +0200)]
alsa-mixer: Fix path set building when using the element-output or element-input mapping options in profile set configuration.
When creating synthesized paths, pa_alsa_path_set_new() created duplicate
elements for each path, and one of the duplicate elements would be marked as
required absent. That made path probing fail. While debugging this, I noticed
also that pa_alsa_path_synthesize() didn't initialize p->last_element properly.
Colin Guthrie [Fri, 25 Feb 2011 10:27:23 +0000 (10:27 +0000)]
core: Add a new hook PA_CORE_HOOK_CARD_PROFILE_CHANGED
This will allow modules to know when a card profile has changed
and take appropriate action. This might prove useful when developing
UCM so that the appropriate verb can be set.
Kim Therkelsen [Fri, 15 Oct 2010 07:25:12 +0000 (09:25 +0200)]
core: Added new hooks: PA_CORE_HOOK_SOURCE_PORT_CHANGED and PA_CORE_HOOK_SINK_PORT_CHANGED
This allows modules to know when certain ports are changed.
This will allow e.g. a filter module (or LADSAP) to only load
when a certain port is used on the device (e.g. to only filter
headphones and not normal speakers).
(Comment from Colin Guthrie: This may also have use in UCM)
Tanu Kaskinen [Mon, 14 Feb 2011 11:41:06 +0000 (13:41 +0200)]
Allow read-only or non-existing sink input volume.
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()).
Tanu Kaskinen [Mon, 7 Feb 2011 16:35:51 +0000 (18:35 +0200)]
core: Link virtual sinks and sources to their streams.
This change doesn't add any functionality in itself, but it will be useful in
the future for operating on chains of sinks or sources that are piggy-backing
on each other.
For example, the PA_PROP_DEVICE_MASTER_DEVICE property could
be handled in the core so that each virtual device doesn't have to maintain it
separately. By using the origin_sink and destination_source pointers the core
is able to see at stream creation time that the stream is created by a virtual
device, and then update that device's property list using the name of the
master device that the stream is being connected to. The same thing can be done
also when the stream is being moved from a device to another, in which case the
_MASTER_DEVICE property needs updating.
Maarten Bosmans [Tue, 4 Jan 2011 16:03:13 +0000 (17:03 +0100)]
Use <pulsecore/socket.h> instead of <sys/socket.h>
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)
Maarten Bosmans [Tue, 4 Jan 2011 10:17:53 +0000 (11:17 +0100)]
Clean up <poll.h> includes
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.
Maarten Bosmans [Wed, 12 Jan 2011 07:15:44 +0000 (08:15 +0100)]
Use setenv instead of putenv
In theory putenv could be used to handle freeing of strings yourself, but this
was not done in PulseAudio. That leaves no advantages in using putenv. With
setenv you're at the mercy of the implementation whether the strings leak, but
at least that is better then a certain leak, as it was before.
Pierre-Louis Bossart [Mon, 31 Jan 2011 19:20:27 +0000 (13:20 -0600)]
alsa: disable period wakeups in tsched mode if possible
This patch reflects a new capability that Lennart was wishing
for. Wish granted...
Re-submitting it now that alsa-lib 1.0.24
provides additional entry points to disable period
wakeups in timer-scheduling mode if hardware can
work without it (HDAudio, oxygen and Intel SST).
Example with standard playback on HDAudio output
Before change:
Top causes for wakeups:
3.8% ( 5.4) [hda_intel] <interrupt>
2.8% ( 4.0) alsa-sink
After change:
Top causes for wakeups:
2.3% ( 3.0) alsa-sink
Arun Raghavan [Tue, 19 Oct 2010 04:29:45 +0000 (09:59 +0530)]
build: Simplify Orc-related make rules
This greatly simplifies the Orc-related make rules. The old system of
distributing generated files is gone, which means that anyone who wants
to build with Orc support enabled needs to have the orcc compiler
installed (presumably the orc 'devel' package in most distros).
Maarten Bosmans [Tue, 25 Jan 2011 10:01:46 +0000 (11:01 +0100)]
build: Generate module symdefs in src/modules directory
This will make it possible to remove the empty Makefile.am files.
- module-...-symdef.h files are all generated in src/modules, instead of in the subdir of the module.
- The default inclusion of src/modules subdirs in AM_CFLAGS can be removed, where necessary (raop) the subdir is
included in the specific CFLAGS.
- The src/daemon and src/modules directories are always created on make, to facilitate out of tree builds.
- AM silent rules are used for the generation of symdef files by m4.
- For echo-cancel, keep the build dir include for now (and mkdir it) although limit it to just the echo-cancel
module's CFLAGS (Colin Guthrie)
David Henningsson [Thu, 9 Dec 2010 10:08:37 +0000 (11:08 +0100)]
Fighting rewinds: Reduce calls to handle_seek
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>
David Henningsson [Mon, 6 Dec 2010 15:25:25 +0000 (16:25 +0100)]
Fighting rewinds: Seek and write data in the same message
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>
Lennart Poettering [Sat, 22 Jan 2011 00:08:36 +0000 (01:08 +0100)]
ratelimit: fix log levels of log suppression messages
When logging a suppression message do so on the same log level as the
suppressed messages.
Cherry picked by Colin Guthrie from ec5a7857127a1b3b9c5517c4a70a9b2c8aab35ca
with a couple of additional changes due to extra limiting in master
that was not present in stable-queue.
Maarten Bosmans [Fri, 7 Jan 2011 00:25:55 +0000 (01:25 +0100)]
Limit rate adjustments to small, inaudible jumps
The same logic is applied to the sample rate adjustments in module-rtp-recv,
module-loopback and module-combine:
- Each time an adjustment is made, the new rate can differ at most 2‰ from the
old rate. Such a step is equal to 3.5 cents (a cent is 1/100th of a
semitone) and as 5 cents is generally considered the smallest observable
difference in pitch, this results in inaudible adjustments.
- The sample rate of the stream can only differ from the rate of the
corresponding sink by 25%. As these adjustments are meant to account for
very small clock drifts, any large deviation from the base rate suggests
something is seriously wrong.
- If the calculated rate is within 20Hz of the base rate, set it to the base
rate. This saves CPU because no resampling is necessary.