You have to post output of alsa-info.sh
Which card is your default sink?
Seem fail with surround21 but does your card suppport multichannel?
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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( 5.156| 0.000) I: [pulseaudio] alsa-util.c: Disabling tsched mode since BATCH
flag is set
( 5.156| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 1981 ms
( 5.158| 0.001) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE
failed (-32)
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Which sst platform are you using?
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/soc/intel/Kconfig
** Changed in: alsa-driver (Ubuntu)
Status: Confirmed => Incomplete
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How did you test ?
When there is hardware low pass filter and downmix widget,
Converter: stream=1, channel=0
Subwoofer channel. Is same as speaker
When there is no hardware low pass filter, channel=1
and you need pulseaudio to provide software lowpass filter and downmixing
Node 0x02
aplay -Dhw:0 --dump-hw-params anystereo.wav
should show channel max is 4 when you need software low pass filter but
only 2 wheb your laptop have hardware low pass filter
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/hda/alc260_quirks.c?id=c29b3f6dd7798964d77199af4925be72a3a48349
This is a regression of removing static model of alc260, you have to
file upstream bug report at bugzilla.kernel.org
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You have to ask the reporter of VGN-A790 laptop which has same pci ssid
104d:81cd as your VGN-A497XP
http://git.alsa-project.org/?p=alsa-
tools.git;a=blob;f=hdajackretask/README;hb=HEAD
You need to find out whether line out node is internal speaker and which
mode is your mic and internal mic
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- /* LINE-2 is used for line-out in rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* select line-out */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LINE-OUT pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* enable HP */
-
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=c3c2c9e7ff3e38bd9ff5b721b6ae8634fce42802
Alc260 basic model was removed, seem use 0x15 as line out and use mono
pin
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Seem same pci ssid
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=5e1b1518a53fc62d9f39a13819c849336c6d8dd4
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snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC260: line_outs=1
(0x10/0x0/0x0/0x0/0x0) type:hp
snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0)
snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0
!!Advanced information - PCI Vendor/Device/Subsystem ID's
!!---
00:1b.0 0403: 8086:2668 (rev 03)
Subsystem: 104d:81cd
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_realtek.c?id=d08c5ef2a039393eaf2ab2152db5f07790fa0f40
You have to retask line out to speaker if node 0x0f is speaker
Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out
Amp-Out caps: ofs=0x00,
Seem similar oops
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1561449
8] WARNING: CPU: 1 PID: 313 at /build/linux-
lAMkDx/linux-4.4.0/drivers/i2c/busses/i2c-designware-baytrail.c:112
baytrail_i2c_acquire+0x139/0x1e0 [i2c_designware_platform]() [ 4.811122]
Modules linked in:
[ 4.588798] WARNING: CPU: 3 PID: 272 at
/build/linux-_Kv5oI/linux-4.2.0/drivers/i2c/busses/i2c-designware-baytrail.c:112
baytrail_i2c_acquire+0x139/0x1e0 [i2c_designware_platform]()
[ 4.588801] Modules linked in: v4l2_common(+) videodev btusb btrtl snd_pcm
media btbcm btintel bluetooth
http://www.cmedia.com.tw/productsdetail/page-p/c1serno-25/c2serno-28/pserno-31.html
8738-LX/MX supports 6CH DAC 5.1 surround output for movies and 3D games.
8738-SX supports 4CH DAC surround output for movies and 3D games.
Built-in 2CH 16bit ADC.
Supports full-duplex 44.1K/48KHz sample rate
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/cmipci.c
CMIPCI_SB_INPUT_SW("Line Capture Route", 4, 3),
You have to ask the Maintainter of cmipci why the control have 4 values if it
emulate sb16
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Mar 21 12:40:41 hostname kernel: usb 3-2: 1:1: usb_set_interface failed (-110)
Mar 21 12:40:41 hostname kernel: usb 3-2: 1:1: usb_set_interface failed (-71)
Mar 21 12:40:41 hostname kernel: usb 3-2: 1:1: usb_set_interface failed (-19)
Seem no volume control
Card hw:4 'VX2'/'TOPPING VX2 at
It seem there is no way to build alsa driver as module after kenerl 4.3
since alsa-driver-build no longer maintained after 2015 october
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You need to open bug report at bugzilla.kernel.org as your snd-cmipci
driver does not have dB ranges and channel map, support 5.1 multi
channels but no volume control for rear center and lfe
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It is unlikely that your patch fix the problem since none of the drivers
of those users have a volume control with multi channel
Only two users have drivers which does not have capture channel map (e.g.
cmipci and intel8x0)
it is not clear why hda-intel can trigger the bug, the acer aspire
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/tree/alsa/gitcompile?h=build#n59
It need aclocal and autoconf to generate configure script
Have you installed the packages?
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https://www.freedesktop.org/software/systemd/man/udevadm.html
Just add
LABEL="pulseaudio_check_usb"
+ATTRS{idVendor}=="1397", ATTRS{idProduct}=="01b0",
ENV{PULSE_IGNORE}="1"
ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1978", ENV{PULSE_PROFILE_SET
}="native-instruments-audio8dj.conf"
udevadm
There is no pcm devices for your card 2 usb midi
Pulseaudio should skip the card Similar to thinkpad ec coonsole at card
29 instead of reporting fail to find working profile
crw-rw+ 1 root audio 116, 2 Mar 18 08:04 /dev/snd/controlC0 crw-
rw+ 1 root audio 116, 12 Mar 18 08:04
sys/class/sound/hwC0D2/init_pin_configs:
0x11 0x40f0
0x12 0x41f0
0x14 0x01014410
0x15 0x01011412
0x16 0x01016411
0x17 0x01012414
0x18 0x01a19c40
0x19 0x02a19c50
0x1a 0x0181344f
0x1b 0x02214c20
0x1c 0x593301f0
0x1d 0x4005e601
0x1e 0x014b6130
0x1f 0x01cb7160
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/core/oss?id=ab30965d9bfcd04931f9b70d00faa2ea614835a8
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https://bugs.launchpad.net/bugs/1551041
Did you change to alsa directory after git checkout build?
The alsa directory should contain
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-build.git/tree/alsa?h=build
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Usually the reason fail to compile are those patch in acore directory
are out dated and you need to report to alsa-devel mailng list
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/tree/alsa/acore?h=build
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!!---Mixer controls for card 1 [U610]
Card hw:1 'U610'/'BEHRINGER UMX 610 at usb-:00:12.0-1, full speed'
Mixer name : 'USB Mixer'
Components : 'USB1397:01b0'
Controls : 0
Simple ctrls : 0
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gitcompile.sh and utils subdirectory
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https://bugs.launchpad.net/bugs/1551041
Title:
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size = 16384,
period size
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/log/?h=build
Last update was 2015 october 12
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https://bugs.launchpad.net/bugs/1551041
https://wiki.ubuntu.com/PulseAudio/Log
https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
Only iec958-stereo-output.conf but no iec958-stereo-input.conf
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Can you post the pulseaudio verbose log?
You have to ask pulseaudio developer whether pulseaudio support spdif
input since your onboard have digital capture device, iec958 capture
switch and capture channel map for device 1
ARECORD List of CAPTURE Hardware Devices
card 1: MID [HDA
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound?id=2d3391ec0ecca37efb6bc995906292f47522b471
The difference between std channel map and alt channel map
You need to modify the driver by adding lines without "+" at the end of the two
routines
+ err =
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/tree/alsa/gitcompile?h=build
The minimum version of the packages are defined at the first few lines
automake
autoconf
libtool
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build.git/log/alsa/gitcompile?h=build
gitcompile only pass the command to configure and run make if no errors
./configure --help
You have to ask the author
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http://git.alsa-project.org/?p=alsa-
lib.git;a=blob;f=src/conf/cards/CMI8738-MC6.conf;hb=HEAD
Seem use different device for front, surround40 and surround51
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/cmipci.c
You need to add channel map in snd_cmipci_pcm_new,
it is strange that pcm_avail.c which only wite one buffer can trigger
when the messages are rate limited
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core/pcm_lib.c?id=74d779ab7c9f9024cfead259206e0e0b20ee37e4
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You have to send email with the errors to Tiwai
The errors related to file in alsa-driver/alsa/acore are related to
changes in sound/core, you need his help to regenerate those patch files
in acore
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/core
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https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules
/module-udev-detect.c
Can you post output of alsa-info.sh for your usb midi?
udev-detect /switch-on-connect need to skip any usb card without pcm
device?
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In theory , there is no need to use any patch in oss directory if you
have disabled oss enumeration by passing --with-oss=no
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/core/oss
You get checkout or use browser to extract the old version of two file
pcm_oss.c and
utils/setup-alsa-kernel -c ../../sound
Only use the following option to build debug driver which enable those dev_dbg
call
--with-debug=full
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The above link is just the oss directory just before
ALSA: oss: underflow in snd_mixer_oss_proc_write(). 2015 october 15
Save the plain text of the two affected files
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[ 9.133753] snd_hda_intel :00:1f.3: azx_get_response timeout,
switching to polling mode: last cmd=0x000f
[ 10.142506] snd_hda_intel :00:1f.3: No response from codec, disabling
MSI: last cmd=0x000f
snd_hda_intel :00:1f.3: Codec #0 probe error; disabling it...
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As Three input jacks of your onboard alc887 were unplugged and
pulseaudio does not regarded CD as available, pulseaudio may switch to
cmi8378 which has no capture channel map
control.40 { iface CARD name 'Rear Mic Jack' value false comment {
access read type BOOLEAN count 1 } } control.41 { iface
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound?id=f49921b894ec338696e47d985acc09df3cddbfd1
Seem no channel map for playback and capture device for your cmi8738, the
playback channel map is device 2
state.CMI8738 {
control.1 { iface PCM device 2 name 'Playback
https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer
/profile-sets/90-pulseaudio.rules
Just add your usb ssid and ENV{PULSE_IGNORE}="1" after those usb devices
You need udev daemom reload the rules or reboot
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A real kernel oops usually consist of stacktrace and dump of all cpu
registers
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https://bugs.launchpad.net/bugs/1551041
Title:
snd_au8820 :01:04.0: BUG: , pos =
You can specify card to just build the needed driver or specify all
cards
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Title:
snd_au8820 :01:04.0: BUG: , pos =
For building alsa driver as module
https://bugs.launchpad.net/ubuntu/+source/alsa-
driver/+bug/1551041/comments/71
You just follow the instruction in that link
What you need to do is
cd alsa-driver
git checkout build
utils/setup-alsa-kernel -c. ../../sound
./gitcompile --enable-dynamic-minors
FIG. 15e is a diagrammatic block diagram of a control register of FIG.
15d, in accordance with one embodiment. Control register 384(c) includes
a sub-buffer enable interrupt flag 384(c)(0), a stop flag 384(c)(1) and
a next segment pointer 384(c)(2). Sub-buffer enable interrupt flag
384(c)(0)
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/refs/tags
You need to know ubuntu 16.04 using which version of kernel
You will find those XRUN/BUGS/invalid position message in system log
before patching and no more message after patching
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You just use any editor to replace this line in au88x0_pcm.c
return (bytes_to_frames(substream->runtime, current_ptr));
by following lines
current_ptr = bytes_to_frames(substream->runtime, current_ptr);
if (current_ptr >= substream->runtime->buffer_size)
current_ptr = 0;
return
https://bugs.launchpad.net/ubuntu/+source/alsa-
driver/+bug/1551041/comments/57
You have to apply the patch manually at the stable ubuntu kernel/driver
and report back to tiwai
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/au88x0?id=54a96dadaa2bbe81006c66447517b57534d80e7f
Before this patch, period_size can be as low as 1 frames,
difference of current page and last page will be 2 or 3 instead of 1
In theory, you should not see the
Sound.git is unstable and not gauarnteed to compile at any time
The safest way is to compile debug kernel/driver at stable ubuntu
release
Tiwai expect you to apply the patch and report back whether it has no side
effect
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/hda_codec.c?id=4b3acaf5b56f53ef259a6ddf8e17dcb529631d9b
It seem that you are using copy front when channel tag of two audio
output are same
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Converter: stream=1, channel=0
The other difference is channel, seem still using two channels when
channels tag of two audio output are same
speaker-test -c4 -t wav -D hw:0,0
only need two channel if there is low pass filter with mono pin complex
and driver don't support 4 channels
For
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core/pcm_lib.c?id=f5914908a5b7b2338f210e56827a1ef35585dc6d
To debug the ptr position at period interrupt, you need alsa driver
build with debug which allow you to enable trace point to log the ptr
position in system
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In theory, you will like to have
Speaker volume and Subwoofer volume
Headphone share with one of them
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Title:
Lenovo
You better post the outout of alsa-info.sh
Since you need to check the amp out and mute of the path from audio
output, audio mixer and pin complex
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The difference between 0x17 mono and 0x1b stereo pin complex
You need to find out whether there is low pass filter when you get sound
from subwoofer and whether both channel affect the subwoofer if the
subwoofer is at the stereo pin complex 0x1b
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Only one pin complex can be retasked at each test since there are only
two audio output/volume control but there are three output pin
complexes, this mean one of the volume control must be shared by two pin
complex
The pin-ctls, power setting of the subwoofer should be same as the
speaker node
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ice1712/ice1712.c?id=4f8e940095536bc002a81666a4107a581c84e9b9
The fix for the other driver in the original bug report
It look like au88x0 pointer callback does not return zero when interrupt occur,
just 2 or 3 frames
dmesg | grep -i "au88"
Since pcm_avail.c only write buffer size once, it is unable to trigger
the bug which ptr > buffer size
so we need to modify the logic to loop more data
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Can you change
periods from 4 to 16
and period_size from 1024 to 256
post the output of pcm_avail.c
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Title:
snd_au8820
Do the number of BUG messsges in system log still 2 frames larger than
buffer size
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Title:
snd_au8820 :01:04.0: BUG: ,
Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps:
ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00]
Connection: 1 0x0d
This seem to be stereo to mono down mix widget since node 0x0d is stereo
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Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right Limits: Playback 0 - 87
Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback
0 [0%] [-65.25dB] [off]
Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch
Node 0x0f is connected to 0x03 through 0x0d
You have to check whether those amp out and mute switfch of the path
from node 3 to node 0x17 is not muted by pulseaudio (e.g. headphone
plasyback volume )
Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker
Playback Volume",
The other output capable pin is node 0x1b, do the driver connected this
node to node 0x02 or 0x03?
Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In
caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00
0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00,
As there is no datasheet, we can only follow the patent description
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/au88x0/au8820.h#n26
FIG. 15c is a diagrammatic block diagram of a mode register 382 of sub-
buffer table 380(3) of FIG. 15b. Mode register 382 includes
Only two of the remain unconnected pins pin-cap support OUT
.0x17 and 0x1b
You can use hdajackretask set one of them as subwoofer, reboot
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https://bugs.freedesktop.org/show_bug.cgi?id=84585#c60
Modified version
You can compare the result
** Bug watch added: freedesktop.org Bugzilla #84585
https://bugs.freedesktop.org/show_bug.cgi?id=84585
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Time diffenece between successive output and frames difference will tell
you whether the card is accurate enoung for pulseaudio timer scheduling
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Can you post the output of pcm_avail.c when using rate 44100 Hz instead
of 48000Hz?
print the time at the same time to check whether the ptr is monotically
increasing as expected rate
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=0ab1ace856205d10cbc1924b2d931c01ffd216a6
This patch only prevent kerneloops pop up, your system log still
flooding with invalid position message every 0.371 second (assume once
per buffer time) when playing audio
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http://mailman.alsa-project.org/pipermail/alsa-
devel/2013-February/059559.html
You need to install many packages in order to build alsa driver
git, autoconf, kernel-header,
You need to rebuild the alsa driver whenever kernel change
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no need to perform git bisect since you already know the reason of
kernel ops is due to "BUG:"
Do ubuntu allow user just build alsa driver as module instead of
recompile the whole kernel
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-build.git/
The oscillator is the silver metal component just above the au8820 chip
of your montego card
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Title:
snd_au8820
You may need to dump mmio VORTEX_SMP_TIME which is tick at 48000 * 256
Hz (what is the clock printed at the oscillator on your au8820 sound
card) to find out the elapsed time between periods is correct or not
- return (bytes_to_frames(substream->runtime, current_ptr));
+ current_ptr =
the differnce between successive output are almost 5 frames
The only bad thing
ptr is not always at period boundary 1024, 2048, 3072 and 4096
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Card numer can be specified by module parameter "index"
Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj :
32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1
beep_mode :
N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable :
Run two alsa applications which using hw device, hwmixvolume allow you
to change the subdevice PCM volume controls which dB range allow gain in
addition to attenuation
For any pulse application, it just control by the subdevice opened by
pulseaudio server
The hardware mixed signal are pass
Each subdevice has its own volume control
control.59 { iface PCM name 'PCM Playback Volume' value.0 0 value.1 0
comment { access 'read write inactive' type INTEGER count 2 range '-128
- 32' dbmin -9600 dbmax 2400 dbvalue.0 0 dbvalue.1 0 } }
control.60 { iface PCM subdevice 1 name 'PCM Playback
There will be volume slider(hardware per stream volume control) for each
application if the application use hw instead of pulse
pavucontrol provide sofware per stream volume control
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His program hardcode to use card 0, you have to change the card number
if ypur au8820 is card 1
or change the order of your two cards by adding specify card number for
the two drivers in /etc/modprobe.d/alsa-base.conf
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You have to send email to the author and commiter since flooding the
system log with message is useless
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https://bugs.launchpad.net/bugs/1551041
Title:
snd_au8820
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core/pcm_lib.c?id=00d90154404ae6218730068d25bf2faad3186631
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Are you the kernel oops is related to au8820 since it just flooding the
system log with BUGS message but no seg fault or dump
it is strange that your bug report did not have pulselist or pulseaudio
log
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Did his program use #include similar to aplay.c?
http://git.alsa-project.org/?p=alsa-
utils.git;a=blob;f=aplay/aplay.c;hb=HEAD
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http://packages.ubuntu.com/search?lang=en=libasound2-dev
For pcm_avail.c
You need to install libasound2-dev and gcc
sudo apt-get install libasound2-dev
gcc pcm_avail.c -o pcm_avail -l asound
./pcm_avail
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Can you post output for your au8820 pci ssid 12eb:0001
do it support D3?
lspci -
[ 0.244875] pci :01:04.0: [12eb:0001] type 00 class 0x040100
[ 0.244898] pci :01:04.0: reg 0x10: [mem 0xfb10-0xfb11]
[ 0.244908] pci :01:04.0: reg 0x14: [io 0x2000-0x2007]
[ 0.244917] pci
[11.080876] snd_au8820 :01:04.0: init started snd_au8820 :01:04.0:
init done.
[ 11.837183] gameport gameport0: AU88x0 Gameport is pci:01:04.0/gameport0,
speed 1087kHz
in theory, you should look for au8820 in system log since those dev_info
also print the dev name
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http://mailman.alsa-project.org/pipermail/alsa-
devel/2014-September/081501.html
you can try Alexander's pcm_avail.c and post the output using your
au8820
It is rather strange that only 2 frames since dma transfer in brust
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dmesg is the command which show the system log similar to
https://launchpadlibrarian.net/243848211/CurrentDmesg.txt
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Title:
You have to recompile alsa driver by adding
if (current_ptr > runtime->buffer_size)
current_ptr = runtime->buffer_size;
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The dirty hack is to ensure current_ptr is less than buffer size in the
above routine
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https://bugs.launchpad.net/bugs/1551041
Title:
snd_au8820 :01:04.0: BUG:
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/au88x0/au88x0_pcm.c
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Title:
snd_au8820
/* pointer callback */
static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream
*substream)
{
vortex_t *chip = snd_pcm_substream_chip(substream);
stream_t *stream = (stream_t *) substream->runtime->private_data;
int dma = stream->dma;
snd_pcm_uframes_t current_ptr = 0;
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