Make sure that the rate plugin does not try to change the slave's
boundary value when setting software parameters; such a change wouldn't
affect the slave pcm.
Furthermore, the stop_threshold conversion code used the wrong boundary
value, and the silence_size conversion code did not take into account
the boundary value at all.
safe_strtoll() now accepts numbers in any base. It formerly assumed that
its input was a decimal number, which had the consequence that
hexadecimal or octal numbers would be parsed as strings when occurring
outside of parameter lists.
This obsoletes some workarounds in the file permission parsing code that
relied on this bug.
This patch fixes a build issue on systems with Gnu assembler
prior to 2.15.92. Instead of having the inline assembler
translate the leal function on a 32 bit register, it is byte encoded.
Signed-off-by: Tobin Davis <tobinx.b.davis@intel.com>
Some io plug-ins might want to adjust the reported delay value and not
strictly follow the current buffer usage (that's why we have two calls
after all).
Allow them to specify a delay() callback and use the previous behaviour
if they don't.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
When this new parameter is specified, the behavior of opening a file PCM
for input is changed to the following: reads from the device, instead of
being passed down to the slave PCM, will read the raw data from the
specified file. No file writes will take place in this case. In the
absence of this parameter, previous behavior is unchanged.
The input file name and descriptor are kept in the snd_pcm_file_t struct
in a manner analogous to the already existing output file parameter.
TODO:
(1) Only interleaved reads (snd_pcm_file_readi) have been implemented
for now.
(2) File read()'s that return a number of bytes that's not a multiple of
the frame size will result in data loss (choppiness). The rbuf,
rbuf_size_bytes, and rbuf_used_bytes members of the snd_pcm_file_t
struct will be used to address this problem in the future.
(3) Mind whether the PCM was opened in blocking mode. If so, we'll have
to loop until the buffer has been filled with read()'s.
Signed-off-by: Juan Carlos Castro y Castro <jcastro@vialink.com.br>
Add the internal start_pending state in the rate plugin for supporting
delayed start in the case no data is committed to slave PCM.
This fixes the problem of CPU hog with artsd.
Fixed the segfault in the error path of snd_pcm_hook_add_conf()
(calling invalid install_func pointer).
Also, added the missing snd_dlclose() to release the handle properly.
The fix to enable finer sync of dmix in delay callback seems
rather harmful for some apps like xmms on SMP.
The problem of aoss was solved by max_periods=0, so better to
revert this change.
Sync pointers in delay callback in dmix. This solves some problems
of aoss with apps that don't check poll_revents.
The rate plugin is fixed to call delay of slave appropriately, too.
Introduce "max_periods" option to specify the max number of periods
per buffer to each plugin.
- When max_periods = -1, the fixed buffer size as the slave size is
used (old behavior).
- When max_periods = 0 (or 1), the number of periods is variable
between 2 and the slave buffer size.
- When max_periods greater than 2 is given, it specifies the max
periods of that pcm explicitly.
When no option is given in the PCM defintion, the value
"defaults.pcm.dmix_max_periods" is referred as default.
The default value is 0, as defined in alsa.conf.
You can override this in ~/.asoundrc or /etc/asound.conf as you like.
Created a new rate converter plugin SDK.
A rate converter can be replaced as an extra plugin now.
The default rate converter is a built-in linear converter.
You can find a sample external converter in alsa-plugins package.
Fixed the noisy output of dmix with two (or less) periods.
The dmix tends to give noise or XRUN when running with two periods
because of its implementation nature. To avoid this, the start
position is aligned to the period size, so that the updates are synced
with interrupts of slave PCM.
Fix the PCM auto-start conditions
- The capture stream should be started the PCM anyway before reads
- Fixed the endless poll() when the written size doesn't fit to the
buffer size due to lack of auto-start
Here's a patch which adds support for S24_3LE and byte-swapped S16 and
S32 to softvol. I've tested S24_3LE and byte-swapped S16 on powerpc
with snd-usb-audio. All other cases are untested so far.
(Config at http://blog.blackdown.de/static/alsa/USB-Audio.conf)
Signed-off-by: Juergen Kreileder <jk@blackdown.de>
snd_timer_tread struct is a bad design for 32/64bit compatibility,
and reading this struct on 32bit program returns zero. This results
in tight poll looping (bug#1938, #1945).
For avoiding this bug, now more bigger buffer is read to cover the
64bit tread struct, too. Also this optimizes the read without
checking -EAGAIN in the case both user-space and kernel have the same
tread size.
With the patch, dmix allows apps to use more flexible buffer sizes.
The max buffer size is unlimited, and the minimal buffer size is
(period size * 2). The buffer size is aligned to period size.
The period size is still bound to the period size of slave PCM.
To back to the old behavior (the fixed buffer size), you can set
defaults.pcm.dmix_variable_buffer false
in your configuration.
Here's a patch for generic dmix which fixes S16 byte swapping.
Tested on powerpc with snd-usb-audio. (Without the patch I get crackling.)
Signed-off-by: Juergen Kreileder <jk@blackdown.de>
Add restriction parameters to pcm hw layer.
The PCM hw has optional parameters, format, rate and channels, to restrict
the configuration. This is useful for definition of surround slave PCMs,
for example.
When calculating the size of the second fragment, do not assume that the
entire size is one period size (which is not true in the draining state)
but use the actual size passed by the caller.
When calculating the continuous part till the end of the buffer, we can
use the slave_frames value that has already been calculated by
snd_pcm_mmap_begin().
