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Issues found by smatch static check: http://smatch.sourceforge.net/
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This prevents overflows and audible artefacts for the audio files which
originally had loudness maximized. Music from audio CD disks is an
example of such files, see http://en.wikipedia.org/wiki/Loudness_war
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Buffer position in X array was not always 16-bytes aligned.
Strict 16-byte alignment is strictly required for powerpc altivec
simd optimizations because altivec does not have support for
unaligned vector loads at all.
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This should make it easier to apply patches from BlueZ which also uses
sbc subdir for this files.
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NREC stands for Noise Reduction and Echo Cancelation, it can be changed
at any point by the headset.
When set to "1" indicates that those algorithms shall be enabled by
default and "0" means the headset probably have them active so they
should be disabled in PA side.
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Use minimum bitpool configured to get the maximum block_size possible,
also remove checks for how much has been written when decoding sbc frames
since the block size may change due to bitpool changes.
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When audio skips it could be that there is some bandwidth limitation in
the link e.g. headset doesn't support EDR (< 2.0), and by reducing
the bitpool it may find a better rate that either prevent the skips
completely or at least reduce them.
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Acquire now return input and output MTU of the file descriptor so it is
no longer necessary to get those after acquiring the fd, which less round
trips and faster response time when switching profiles.
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The ORCC stage does this anyway (and this was buggy anyway as it had an extra
'src/' prefix so never worked properly when it was introduced in d6cdd80).
Thanks to Maarten Bosmans for pointing it out.
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already corked.
Although by "cork" I really mean "cork+mute" as that's what the module
does.
If e.g. Rhythmbox is paused when a phone call comes in, the current
stream state will be corked and thus we should not track it for future
uncorking when the phone call ends.
Likewise if the stream is just muted (manually) we will not take any
action either when the phone stream is seen first, nor when it
disappears.
Also add some additional debug messages.
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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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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.
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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.
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pa_sink_set_volume() and friends accept mono volumes too, so no need to impose
unneeded restrictions in the D-Bus API.
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complain about unhandled cases.
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This commit only implements the parser, the decibel fix data is not yet used
for anything.
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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.
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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).
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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.
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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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Also fix a left over reference to HAL in default.pa
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This completes the client-side changes to the protocol extension
introduced by commit 99ddca89cdca9b0b92ab9870764f9211e6a82e31
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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.
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(part of a patch series removing all ramping code)
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audio signals"
This reverts commit 95a98fe6f2002c9dd448b70bb6944541b5616df3.
Conflicts:
src/Makefile.am
src/pulsecore/envelope.c
src/pulsecore/envelope.h
src/tests/envelope-test.c
(part of a patch series removing all ramping code)
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This reverts commit 8eaa40b6f4cae749610770c85ba500f326d59b50.
Conflicts:
src/pulsecore/envelope.c
(part of a patch series removing all ramping code)
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This reverts commit 5318eb35ef3f91836084382a4f3d5ef08d322554.
Conflicts:
src/pulsecore/sink-input.c
(part of a patch series removing all ramping code)
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This reverts commit 897ef86b7fbb87ef17d30c584e6cd93abfc342bc.
Conflicts:
src/pulsecore/sink.c
(part of a patch series removing all ramping code)
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This reverts commit f202af17b717f5b383ac072f80a6c1327bc3143b.
(part of a patch series removing all ramping code)
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This reverts commit aa9348441db34b787784711f19882d6c42fa315d.
(part of a patch series removing all ramping code)
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Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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Also remove src/module/.gitignore as this is no longer needed
as pointed out by Arun Raghavan
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If send_msg is false, the message will be sent by the caller.
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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.
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