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Merge branch 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa 

* 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (264 commits)
  [ALSA] version 1.0.15
  [ALSA] Fix thinko in cs4231 mce down check
  [ALSA] sun-cs4231: improved waiting after MCE down
  [ALSA] sun-cs4231: use cs4231-regs.h
  [ALSA] This simplifies and fixes waiting loops of the mce_down()
  [ALSA] This patch adds support for a wavetable chip on
  [ALSA] This patch removes open_mutex from the ad1848-lib as
  [ALSA] fix bootup crash in snd_gus_interrupt()
  [ALSA] hda-codec - Fix SKU ID function for realtek codecs
  [ALSA] Support  ASUS P701 eeepc [0x1043 0x82a1] support
  [ALSA] hda-codec - Add array terminator for dmic in STAC codec
  [ALSA] hdsp - Fix zero division
  [ALSA] usb-audio - Fix double comment
  [ALSA] hda-codec - Fix STAC922x volume knob control
  [ALSA] Changed Jaroslav Kysela's e-mail from perex@suse.cz to perex@perex.cz
  [ALSA] hda-codec - Fix for Fujitsu Lifebook C1410
  [ALSA] mpu-401: remove MPU401_INFO_UART_ONLY flag
  [ALSA] mpu-401: do not require an ACK byte for the ENTER_UART command
  [ALSA] via82xx - Add DXS quirk for Shuttle AK31v2
  [ALSA] hda-codec - Fix input_mux numbers for vaio stac92xx
  ...
hifive-unleashed-5.1
Linus Torvalds 2007-10-16 10:13:38 -07:00
parent 92d15c2ccb 24837e6f24
commit fc8a327db6
370 changed files with 11195 additions and 5980 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
CREDITS
View File

@ -1933,7 +1933,7 @@ M: seasons@makosteszta.sote.hu
D: Original author of software suspend
N: Jaroslav Kysela
E: perex@suse.cz
E: perex@perex.cz
W: http://www.perex.cz
D: Original Author and Maintainer for HP 10/100 Mbit Network Adapters
D: ISA PnP

115
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -365,13 +365,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module snd-cmipci
-----------------
Module for C-Media CMI8338 and 8738 PCI sound cards.
Module for C-Media CMI8338/8738/8768/8770 PCI sound cards.
mpu_port - 0x300,0x310,0x320,0x330 = legacy port,
1 = integrated PCI port,
mpu_port - port address of MIDI interface (8338 only):
0x300,0x310,0x320,0x330 = legacy port,
0 = disable (default)
fm_port - 0x388 = legacy port,
1 = integrated PCI port (default),
fm_port - port address of OPL-3 FM synthesizer (8x38 only):
0x388 = legacy port,
1 = integrated PCI port (default on 8738),
0 = disable
soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only)
(default = 1)
@ -768,6 +769,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
single_cmd - Use single immediate commands to communicate with
codecs (for debugging only)
enable_msi - Enable Message Signaled Interrupt (MSI) (default = off)
power_save - Automatic power-saving timtout (in second, 0 =
disable)
power_save_controller - Reset HD-audio controller in power-saving mode
(default = on)
This module supports one card and autoprobe.
@ -828,6 +833,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
ALC268
3stack 3-stack model
toshiba Toshiba A205
acer Acer laptops
auto auto-config reading BIOS (default)
ALC662
@ -842,7 +849,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
arima Arima W820Di1
targa Targa T8, MSI-1049 T8
asus-a7j ASUS A7J
asus-a7m ASUS A7M
macpro MacPro support
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
auto auto-config reading BIOS (default)
@ -854,6 +865,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
6stack-dig-demo 6-jack digital for Intel demo board
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
medion Medion Laptops
medion-md2 Medion MD2
targa-dig Targa/MSI
@ -862,6 +874,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
lenovo-101e Lenovo 101E
lenovo-nb0763 Lenovo NB0763
lenovo-ms7195-dig Lenovo MS7195
haier-w66 Haier W66
6stack-hp HP machines with 6stack (Nettle boards)
3stack-hp HP machines with 3stack (Lucknow, Samba boards)
auto auto-config reading BIOS (default)
@ -885,6 +898,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
lenovo Lenovo 3000 C200
dallas Dallas laptops
hp HP TX1000
auto auto-config reading BIOS (default)
CMI9880
@ -920,6 +934,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack 3-stack, shared surrounds
laptop 2-channel only (FSC V2060, Samsung M50)
laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J)
laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
ultra 2-channel with EAPD (Samsung Ultra tablet PC)
AD1988
@ -945,14 +960,30 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
STAC9200/9205/9254
STAC9200
ref Reference board
dell-d21 Dell (unknown)
dell-d22 Dell (unknown)
dell-d23 Dell (unknown)
dell-m21 Dell Inspiron 630m, Dell Inspiron 640m
dell-m22 Dell Latitude D620, Dell Latitude D820
dell-m23 Dell XPS M1710, Dell Precision M90
dell-m24 Dell Latitude 120L
dell-m25 Dell Inspiron E1505n
dell-m26 Dell Inspiron 1501
dell-m27 Dell Inspiron E1705/9400
gateway Gateway laptops with EAPD control
STAC9205/9254
ref Reference board
dell-m42 Dell (unknown)
dell-m43 Dell Precision
dell-m44 Dell Inspiron
STAC9220/9221
ref Reference board
3stack D945 3stack
5stack D945 5stack + SPDIF
dell Dell XPS M1210
intel-mac-v1 Intel Mac Type 1
intel-mac-v2 Intel Mac Type 2
intel-mac-v3 Intel Mac Type 3
@ -964,6 +995,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3)
imac-intel Intel iMac (eq. type 2)
imac-intel-20 Intel iMac (newer version) (eq. type 3)
dell-d81 Dell (unknown)
dell-d82 Dell (unknown)
dell-m81 Dell (unknown)
dell-m82 Dell XPS M1210
STAC9202/9250/9251
ref Reference board, base config
@ -975,6 +1010,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
ref Reference board
3stack D965 3stack
5stack D965 5stack + SPDIF
dell-3stack Dell Dimension E520
STAC9872
vaio Setup for VAIO FE550G/SZ110
@ -989,6 +1025,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
ML (see the section "Links and Addresses").
power_save and power_save_controller options are for power-saving
mode. See powersave.txt for details.
Note 2: If you get click noises on output, try the module option
position_fix=1 or 2. position_fix=1 will use the SD_LPIB
register value without FIFO size correction as the current
@ -1349,7 +1388,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
port - port number or -1 (disable)
irq - IRQ number or -1 (disable)
pnp - PnP detection - 0 = disable, 1 = enable (default)
uart_enter - Issue UART_ENTER command at open - bool, default = on
This module supports multiple devices and PnP.
@ -1630,6 +1668,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
The power-management is supported.
Module snd-sc6000
-----------------
Module for Gallant SC-6000 soundcard.
port - Port # (0x220 or 0x240)
mss_port - MSS Port # (0x530 or 0xe80)
irq - IRQ # (5,7,9,10,11)
mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq
dma - DMA # (1,3,0)
This module supports multiple cards.
This card is also known as Audio Excel DSP 16 or Zoltrix AV302.
Module snd-sgalaxy
------------------
@ -1650,9 +1703,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for ENSONIQ SoundScape PnP cards.
port - Port # (PnP setup)
wss_port - WSS Port # (PnP setup)
irq - IRQ # (PnP setup)
mpu_irq - MPU-401 IRQ # (PnP setup)
dma - DMA # (PnP setup)
dma2 - 2nd DMA # (PnP setup, -1 to disable)
This module supports multiple cards. ISA PnP must be enabled.
You need sscape_ctl tool in alsa-tools package for loading
@ -1697,8 +1752,52 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
dma2 - DMA2 # for CS4232 PCM interface.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
The below are options for wavefront_synth features:
wf_raw - Assume that we need to boot the OS (default:no)
If yes, then during driver loading, the state of the board is
ignored, and we reset the board and load the firmware anyway.
fx_raw - Assume that the FX process needs help (default:yes)
If false, we'll leave the FX processor in whatever state it is
when the driver is loaded. The default is to download the
microprogram and associated coefficients to set it up for
"default" operation, whatever that means.
debug_default - Debug parameters for card initialization
wait_usecs - How long to wait without sleeping, usecs
(default:150)
This magic number seems to give pretty optimal throughput
based on my limited experimentation.
If you want to play around with it and find a better value, be
my guest. Remember, the idea is to get a number that causes us
to just busy wait for as many WaveFront commands as possible,
without coming up with a number so large that we hog the whole
CPU.
Specifically, with this number, out of about 134,000 status
waits, only about 250 result in a sleep.
sleep_interval - How long to sleep when waiting for reply
(default: 100)
sleep_tries - How many times to try sleeping during a wait
(default: 50)
ospath - Pathname to processed ICS2115 OS firmware
(default:wavefront.os)
The path name of the ISC2115 OS firmware. In the recent
version, it's handled via firmware loader framework, so it
must be installed in the proper path, typically,
/lib/firmware.
reset_time - How long to wait for a reset to take effect
(default:2)
ramcheck_time - How many seconds to wait for the RAM test
(default:20)
osrun_time - How many seconds to wait for the ICS2115 OS
(default:10)
This module supports multiple cards and ISA PnP.
Note: the firmware file "wavefront.os" was located in the earlier
version in /etc. Now it's loaded via firmware loader, and
must be in the proper firmware path, such as /lib/firmware.
Copy (or symlink) the file appropriately if you get an error
regarding firmware downloading after upgrading the kernel.
Module snd-sonicvibes
---------------------

17
Documentation/sound/alsa/CMIPCI.txt
View File

@ -1,5 +1,5 @@
Brief Notes on C-Media 8738/8338 Driver
=======================================
Brief Notes on C-Media 8338/8738/8768/8770 Driver
=================================================
Takashi Iwai <tiwai@suse.de>
@ -209,10 +209,13 @@ In addition to the standard SB mixer, CM8x38 provides more functions.
MIDI CONTROLLER
---------------
The MPU401-UART interface is disabled as default. You need to set
module option "mpu_port" with a valid I/O port address to enable the
MIDI support. The valid I/O ports are 0x300, 0x310, 0x320 and 0x330.
Choose the value which doesn't conflict with other cards.
With CMI8338 chips, the MPU401-UART interface is disabled as default.
You need to set the module option "mpu_port" to a valid I/O port address
to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and
0x330. Choose a value that doesn't conflict with other cards.
With CMI8738 and newer chips, the MIDI interface is enabled by default
and the driver automatically chooses a port address.
There is _no_ hardware wavetable function on this chip (except for
OPL3 synth below).
@ -230,6 +233,8 @@ Set "fm_port" module option for more cards.
The output quality of FM OPL/3 is, however, very weird.
I don't know why..
CMI8768 and newer chips do not have the FM synth.
Joystick and Modem
------------------

184
Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
View File

@ -18,8 +18,8 @@
</affiliation>
</author>
<date>November 17, 2005</date>
<edition>0.3.6</edition>
<date>September 10, 2007</date>
<edition>0.3.7</edition>
<abstract>
<para>
@ -405,8 +405,9 @@
/* definition of the chip-specific record */
struct mychip {
struct snd_card *card;
// rest of implementation will be in the section
// "PCI Resource Managements"
/* rest of implementation will be in the section
* "PCI Resource Managements"
*/
};
/* chip-specific destructor
@ -414,7 +415,7 @@
*/
static int snd_mychip_free(struct mychip *chip)
{
.... // will be implemented later...
.... /* will be implemented later... */
}
/* component-destructor
@ -440,8 +441,9 @@
*rchip = NULL;
// check PCI availability here
// (see "PCI Resource Managements")
/* check PCI availability here
* (see "PCI Resource Managements")
*/
....
/* allocate a chip-specific data with zero filled */
@ -451,12 +453,13 @@
chip->card = card;
// rest of initialization here; will be implemented
// later, see "PCI Resource Managements"
/* rest of initialization here; will be implemented
* later, see "PCI Resource Managements"
*/
....
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
chip, &ops)) < 0) {
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
snd_mychip_free(chip);
return err;
}
@ -490,7 +493,8 @@
return -ENOMEM;
/* (3) */
if ((err = snd_mychip_create(card, pci, &chip)) < 0) {
err = snd_mychip_create(card, pci, &chip);
if (err < 0) {
snd_card_free(card);
return err;
}
@ -502,10 +506,11 @@
card->shortname, chip->ioport, chip->irq);
/* (5) */
.... // implemented later
.... /* implemented later */
/* (6) */
if ((err = snd_card_register(card)) < 0) {
err = snd_card_register(card);
if (err < 0) {
snd_card_free(card);
return err;
}
@ -605,7 +610,8 @@
<![CDATA[
struct mychip *chip;
....
if ((err = snd_mychip_create(card, pci, &chip)) < 0) {
err = snd_mychip_create(card, pci, &chip);
if (err < 0) {
snd_card_free(card);
return err;
}
@ -666,7 +672,8 @@
<informalexample>
<programlisting>
<![CDATA[
if ((err = snd_card_register(card)) < 0) {
err = snd_card_register(card);
if (err < 0) {
snd_card_free(card);
return err;
}
@ -1091,7 +1098,7 @@
static int snd_mychip_free(struct mychip *chip)
{
/* disable hardware here if any */
.... // (not implemented in this document)
.... /* (not implemented in this document) */
/* release the irq */
if (chip->irq >= 0)
@ -1119,7 +1126,8 @@
*rchip = NULL;
/* initialize the PCI entry */
if ((err = pci_enable_device(pci)) < 0)
err = pci_enable_device(pci);
if (err < 0)
return err;
/* check PCI availability (28bit DMA) */
if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
@ -1141,7 +1149,8 @@
chip->irq = -1;
/* (1) PCI resource allocation */
if ((err = pci_request_regions(pci, "My Chip")) < 0) {
err = pci_request_regions(pci, "My Chip");
if (err < 0) {
kfree(chip);
pci_disable_device(pci);
return err;
@ -1156,10 +1165,10 @@
chip->irq = pci->irq;
/* (2) initialization of the chip hardware */
.... // (not implemented in this document)
.... /* (not implemented in this document) */
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
chip, &ops)) < 0) {
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
snd_mychip_free(chip);
return err;
}
@ -1233,7 +1242,8 @@
<informalexample>
<programlisting>
<![CDATA[
if ((err = pci_enable_device(pci)) < 0)
err = pci_enable_device(pci);
if (err < 0)
return err;
if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
@ -1294,7 +1304,8 @@
<informalexample>
<programlisting>
<![CDATA[
if ((err = pci_request_regions(pci, "My Chip")) < 0) {
err = pci_request_regions(pci, "My Chip");
if (err < 0) {
kfree(chip);
pci_disable_device(pci);
return err;
@ -1322,7 +1333,7 @@
<programlisting>
<![CDATA[
if (request_irq(pci->irq, snd_mychip_interrupt,
IRQF_DISABLED|IRQF_SHARED, "My Chip", chip)) {
IRQF_SHARED, "My Chip", chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
@ -1773,7 +1784,8 @@
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_mychip_playback_hw;
// more hardware-initialization will be done here
/* more hardware-initialization will be done here */
....
return 0;
}
@ -1781,7 +1793,8 @@
static int snd_mychip_playback_close(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
// the hardware-specific codes will be here
/* the hardware-specific codes will be here */
....
return 0;
}
@ -1793,7 +1806,8 @@
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_mychip_capture_hw;
// more hardware-initialization will be done here
/* more hardware-initialization will be done here */
....
return 0;
}
@ -1801,7 +1815,8 @@
static int snd_mychip_capture_close(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
// the hardware-specific codes will be here
/* the hardware-specific codes will be here */
....
return 0;
}
@ -1844,10 +1859,12 @@
{
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
// do something to start the PCM engine
/* do something to start the PCM engine */
....
break;
case SNDRV_PCM_TRIGGER_STOP:
// do something to stop the PCM engine
/* do something to stop the PCM engine */
....
break;
default:
return -EINVAL;
@ -1900,8 +1917,8 @@
struct snd_pcm *pcm;
int err;
if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1,
&pcm)) < 0)
err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
strcpy(pcm->name, "My Chip");
@ -1939,8 +1956,8 @@
struct snd_pcm *pcm;
int err;
if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1,
&pcm)) < 0)
err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
strcpy(pcm->name, "My Chip");
@ -2097,7 +2114,7 @@
struct mychip *chip = snd_pcm_chip(pcm);
/* free your own data */
kfree(chip->my_private_pcm_data);
// do what you like else
/* do what you like else */
....
}
@ -2884,10 +2901,10 @@ struct _snd_pcm_runtime {
<![CDATA[
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
// do something to start the PCM engine
/* do something to start the PCM engine */
break;
case SNDRV_PCM_TRIGGER_STOP:
// do something to stop the PCM engine
/* do something to stop the PCM engine */
break;
default:
return -EINVAL;
@ -3071,7 +3088,7 @@ struct _snd_pcm_runtime {
spin_unlock(&chip->lock);
snd_pcm_period_elapsed(chip->substream);
spin_lock(&chip->lock);
// acknowledge the interrupt if necessary
/* acknowledge the interrupt if necessary */
}
....
spin_unlock(&chip->lock);
@ -3134,7 +3151,7 @@ struct _snd_pcm_runtime {
snd_pcm_period_elapsed(substream);
spin_lock(&chip->lock);
}
// acknowledge the interrupt if necessary
/* acknowledge the interrupt if necessary */
}
....
spin_unlock(&chip->lock);
@ -3455,6 +3472,13 @@ struct _snd_pcm_runtime {
(casted to unsigned long) of some record to this field, too.
</para>
<para>
The <structfield>tlv</structfield> field can be used to provide
metadata about the control; see the
<link linkend="control-interface-tlv">
<citetitle>Metadata</citetitle></link> subsection.
</para>
<para>
The other three are
<link linkend="control-interface-callbacks"><citetitle>
@ -3604,7 +3628,7 @@ struct _snd_pcm_runtime {
<title>Example of info callback</title>
<programlisting>
<![CDATA[
static int snd_myctl_info(struct snd_kcontrol *kcontrol,
static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
@ -3639,7 +3663,7 @@ struct _snd_pcm_runtime {
<informalexample>
<programlisting>
<![CDATA[
static int snd_myctl_info(struct snd_kcontrol *kcontrol,
static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[4] = {
@ -3658,6 +3682,16 @@ struct _snd_pcm_runtime {
</programlisting>
</informalexample>
</para>
<para>
Some common info callbacks are prepared for easy use:
<function>snd_ctl_boolean_mono_info()</function> and
<function>snd_ctl_boolean_stereo_info()</function>.
Obviously, the former is an info callback for a mono channel
boolean item, just like <function>snd_myctl_mono_info</function>
above, and the latter is for a stereo channel boolean item.
</para>
</section>
<section id="control-interface-callbacks-get">
@ -3794,7 +3828,8 @@ struct _snd_pcm_runtime {
<informalexample>
<programlisting>
<![CDATA[
if ((err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip))) < 0)
err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip));
if (err < 0)
return err;
]]>
</programlisting>
@ -3843,6 +3878,56 @@ struct _snd_pcm_runtime {
</para>
</section>
<section id="control-interface-tlv">
<title>Metadata</title>
<para>
To provide information about the dB values of a mixer control, use
on of the <constant>DECLARE_TLV_xxx</constant> macros from
<filename>&lt;sound/tlv.h&gt;</filename> to define a variable
containing this information, set the<structfield>tlv.p
</structfield> field to point to this variable, and include the
<constant>SNDRV_CTL_ELEM_ACCESS_TLV_READ</constant> flag in the
<structfield>access</structfield> field; like this:
<informalexample>
<programlisting>
<![CDATA[
static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0);
static struct snd_kcontrol_new my_control __devinitdata = {
...
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
SNDRV_CTL_ELEM_ACCESS_TLV_READ,
...
.tlv.p = db_scale_my_control,
};
]]>
</programlisting>
</informalexample>
</para>
<para>
The <function>DECLARE_TLV_DB_SCALE</function> macro defines
information about a mixer control where each step in the control's
value changes the dB value by a constant dB amount.
The first parameter is the name of the variable to be defined.
The second parameter is the minimum value, in units of 0.01 dB.
The third parameter is the step size, in units of 0.01 dB.
Set the fourth parameter to 1 if the minimum value actually mutes
the control.
</para>
<para>
The <function>DECLARE_TLV_DB_LINEAR</function> macro defines
information about a mixer control where the control's value affects
the output linearly.
The first parameter is the name of the variable to be defined.
The second parameter is the minimum value, in units of 0.01 dB.
The third parameter is the maximum value, in units of 0.01 dB.
If the minimum value mutes the control, set the second parameter to
<constant>TLV_DB_GAIN_MUTE</constant>.
</para>
</section>
</chapter>
@ -3880,7 +3965,7 @@ struct _snd_pcm_runtime {
{
struct mychip *chip = ac97->private_data;
....
// read a register value here from the codec
/* read a register value here from the codec */
return the_register_value;
}
@ -3889,7 +3974,7 @@ struct _snd_pcm_runtime {
{
struct mychip *chip = ac97->private_data;
....
// write the given register value to the codec
/* write the given register value to the codec */
}
static int snd_mychip_ac97(struct mychip *chip)
@ -3902,7 +3987,8 @@ struct _snd_pcm_runtime {
.read = snd_mychip_ac97_read,
};
if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0)
err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus);
if (err < 0)
return err;
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = chip;
@ -4447,10 +4533,10 @@ struct _snd_pcm_runtime {
<informalexample>
<programlisting>
<![CDATA[
struct list_head *list;
struct snd_rawmidi_substream *substream;
list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
substream = list_entry(list, struct snd_rawmidi_substream, list);
list_for_each_entry(substream,
&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams,
list {
sprintf(substream->name, "My MIDI Port %d", substream->number + 1);
}
/* same for SNDRV_RAWMIDI_STREAM_INPUT */

7
Documentation/sound/alsa/OSS-Emulation.txt
View File

@ -303,10 +303,3 @@ ICE1712 supports only the unconventional format, interleaved
the buffer as the conventional (mono or 2-channels, 8 or 16bit) format
on OSS.
USB devices
-----------
Some USB devices support only 24bit format packed in 3bytes. This
format is not supported by OSS and no conversion is provided by kernel
OSS emulation. You can use the user-space OSS emulation via libaoss
instead.

49
Documentation/sound/alsa/hda_codec.txt
View File

@ -49,6 +49,9 @@ struct hda_bus_ops {
unsigned int verb, unsigned int parm);
unsigned int (*get_response)(struct hda_codec *codec);
void (*private_free)(struct hda_bus *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
void (*pm_notify)(struct hda_codec *codec);
#endif
};
The command callback is called when the codec module needs to send a
@ -56,9 +59,16 @@ VERB to the controller. It's always a single command.
The get_response callback is called when the codec requires the answer
for the last command. These two callbacks are mandatory and have to
be given.
The last, private_free callback, is optional. It's called in the
The third, private_free callback, is optional. It's called in the
destructor to release any necessary data in the lowlevel driver.
The pm_notify callback is available only with
CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs
to power up or may power down. The controller should check the all
belonging codecs on the bus whether they are actually powered off
(check codec->power_on), and optionally the driver may power down the
contoller side, too.
The bus instance is created via snd_hda_bus_new(). You need to pass
the card instance, the template, and the pointer to store the
resultant bus instance.
@ -86,10 +96,8 @@ resultant codec instance (can be NULL if not needed).
The codec is stored in a linked list of bus instance. You can follow
the codec list like:
struct list_head *p;
struct hda_codec *codec;
list_for_each(p, &bus->codec_list) {
codec = list_entry(p, struct hda_codec, list);
list_for_each_entry(codec, &bus->codec_list, list) {
...
}
@ -100,10 +108,15 @@ initialization sequence is called when the controls are built later.
Codec Access
============
To access codec, use snd_codec_read() and snd_codec_write().
To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
snd_hda_param_read() is for reading parameters.
For writing a sequence of verbs, use snd_hda_sequence_write().
There are variants of cached read/write, snd_hda_codec_write_cache(),
snd_hda_sequence_write_cache(). These are used for recording the
register states for the power-mangement resume. When no PM is needed,
these are equivalent with non-cached version.
To retrieve the number of sub nodes connected to the given node, use
snd_hda_get_sub_nodes(). The connection list can be obtained via
snd_hda_get_connections() call.
@ -239,6 +252,10 @@ set the codec->patch_ops field. This is defined as below:
int (*suspend)(struct hda_codec *codec, pm_message_t state);
int (*resume)(struct hda_codec *codec);
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
int (*check_power_status)(struct hda_codec *codec,
hda_nid_t nid);
#endif
};
The build_controls callback is called from snd_hda_build_controls().
@ -251,6 +268,18 @@ The unsol_event callback is called when an unsolicited event is
received.
The suspend and resume callbacks are for power management.
They can be NULL if no special sequence is required. When the resume
callback is NULL, the driver calls the init callback and resumes the
registers from the cache. If other handling is needed, you'd need to
write your own resume callback. There, the amp values can be resumed
via
void snd_hda_codec_resume_amp(struct hda_codec *codec);
and the other codec registers via
void snd_hda_codec_resume_cache(struct hda_codec *codec);
The check_power_status callback is called when the amp value of the
given widget NID is changed. The codec code can turn on/off the power
appropriately from this information.
Each entry can be NULL if not necessary to be called.
@ -267,8 +296,7 @@ Digital I/O
===========
Call snd_hda_create_spdif_out_ctls() from the patch to create controls
related with SPDIF out. In the patch resume callback, call
snd_hda_resume_spdif().
related with SPDIF out.
Helper Functions
@ -284,12 +312,7 @@ as a module parameter, and PCI subsystem IDs. If the matching entry
is found, it returns the config field value.
snd_hda_add_new_ctls() can be used to create and add control entries.
Pass the zero-terminated array of struct snd_kcontrol_new. The same array
can be passed to snd_hda_resume_ctls() for resume.
Note that this will call control->put callback of these entries. So,
put callback should check codec->in_resume and force to restore the
given value if it's non-zero even if the value is identical with the
cached value.
Pass the zero-terminated array of struct snd_kcontrol_new
Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
used for the entry of struct snd_kcontrol_new.

41
Documentation/sound/alsa/powersave.txt 100644
View File

@ -0,0 +1,41 @@
Notes on Power-Saving Mode
==========================
AC97 and HD-audio drivers have the automatic power-saving mode.
This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE
and CONFIG_SND_HDA_POWER_SAVE options, respectively.
With the automatic power-saving, the driver turns off the codec power
appropriately when no operation is required. When no applications use
the device and/or no analog loopback is set, the power disablement is
done fully or partially. It'll save a certain power consumption, thus
good for laptops (even for desktops).
The time-out for automatic power-off can be specified via power_save
module option of snd-ac97-codec and snd-hda-intel modules. Specify
the time-out value in seconds. 0 means to disable the automatic
power-saving. The default value of timeout is given via
CONFIG_SND_AC97_POWER_SAVE_DEFAULT and
CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1
(the minimum value) isn't recommended because many applications try to
reopen the device frequently. 10 would be a good choice for normal
operations.
The power_save option is exported as writable. This means you can
adjust the value via sysfs on the fly. For example, to turn on the
automatic power-save mode with 10 seconds, write to
/sys/modules/snd_ac97_codec/parameters/power_save (usually as root):
# echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save
Note that you might hear click noise/pop when changing the power
state. Also, it often takes certain time to wake up from the
power-down to the active state. These are often hardly to fix, so
don't report extra bug reports unless you have a fix patch ;-)
For HD-audio interface, there is another module option,
power_save_controller. This enables/disables the power-save mode of
the controller side. Setting this on may reduce a bit more power
consumption, but might result in longer wake-up time and click noise.
Try to turn it off when you experience such a thing too often.

6
MAINTAINERS
View File

@ -1769,7 +1769,7 @@ S: Maintained
HP100: Driver for HP 10/100 Mbit/s Voice Grade Network Adapter Series
P: Jaroslav Kysela
M: perex@suse.cz
M: perex@perex.cz
S: Maintained
HPET: High Precision Event Timers driver (hpet.c)
@ -2132,7 +2132,7 @@ S: Maintained
ISAPNP
P: Jaroslav Kysela
M: perex@suse.cz
M: perex@perex.cz
S: Maintained
ISDN SUBSYSTEM
@ -3523,7 +3523,7 @@ S: Maintained
SOUND
P: Jaroslav Kysela
M: perex@suse.cz
M: perex@perex.cz
L: alsa-devel@alsa-project.org (subscribers-only)
S: Maintained

2
drivers/media/video/cx88/cx88-alsa.c
View File

@ -7,7 +7,7 @@
* (c) 2005,2006 Ricardo Cerqueira <v4l@cerqueira.org>
* (c) 2005 Mauro Carvalho Chehab <mchehab@infradead.org>
* Based on a dummy cx88 module by Gerd Knorr <kraxel@bytesex.org>
* Based on dummy.c by Jaroslav Kysela <perex@suse.cz>
* Based on dummy.c by Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

4
drivers/net/hp100.c
View File

@ -8,7 +8,7 @@
** Extended for new busmaster capable chipsets by
** Siegfried "Frieder" Loeffler (dg1sek) <floeff@mathematik.uni-stuttgart.de>
**
** Maintained by: Jaroslav Kysela <perex@suse.cz>
** Maintained by: Jaroslav Kysela <perex@perex.cz>
**
** This driver has only been tested with
** -- HP J2585B 10/100 Mbit/s PCI Busmaster
@ -2951,7 +2951,7 @@ static struct pci_driver hp100_pci_driver = {
*/
MODULE_LICENSE("GPL");
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, "
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, "
"Siegfried \"Frieder\" Loeffler (dg1sek) <floeff@mathematik.uni-stuttgart.de>");
MODULE_DESCRIPTION("HP CASCADE Architecture Driver for 100VG-AnyLan Network Adapters");

2
drivers/pnp/interface.c
View File

@ -1,7 +1,7 @@
/*
* interface.c - contains everything related to the user interface
*
* Some code, especially possible resource dumping is based on isapnp_proc.c (c) Jaroslav Kysela <perex@suse.cz>
* Some code, especially possible resource dumping is based on isapnp_proc.c (c) Jaroslav Kysela <perex@perex.cz>
* Copyright 2002 Adam Belay <ambx1@neo.rr.com>
*/

4
drivers/pnp/isapnp/core.c
View File

@ -1,6 +1,6 @@
/*
* ISA Plug & Play support
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -53,7 +53,7 @@ static int isapnp_rdp; /* Read Data Port */
static int isapnp_reset = 1; /* reset all PnP cards (deactivate) */
static int isapnp_verbose = 1; /* verbose mode */
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Generic ISA Plug & Play support");
module_param(isapnp_disable, int, 0);
MODULE_PARM_DESC(isapnp_disable, "ISA Plug & Play disable");

2
drivers/pnp/isapnp/proc.c
View File

@ -1,6 +1,6 @@
/*
* ISA Plug & Play support
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

2
drivers/pnp/manager.c
View File

@ -1,7 +1,7 @@
/*
* manager.c - Resource Management, Conflict Resolution, Activation and Disabling of Devices
*
* based on isapnp.c resource management (c) Jaroslav Kysela <perex@suse.cz>
* based on isapnp.c resource management (c) Jaroslav Kysela <perex@perex.cz>
* Copyright 2003 Adam Belay <ambx1@neo.rr.com>
*/

2
drivers/pnp/resource.c
View File

@ -1,7 +1,7 @@
/*
* resource.c - Contains functions for registering and analyzing resource information
*
* based on isapnp.c resource management (c) Jaroslav Kysela <perex@suse.cz>
* based on isapnp.c resource management (c) Jaroslav Kysela <perex@perex.cz>
* Copyright 2003 Adam Belay <ambx1@neo.rr.com>
*/

1
include/linux/i2c-id.h
View File

@ -120,6 +120,7 @@
#define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */
#define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */
#define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */
#define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */
#define I2C_DRIVERID_I2CDEV 900
#define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */

25
include/linux/spi/at73c213.h 100644
View File

@ -0,0 +1,25 @@
/*
* Board-specific data used to set up AT73c213 audio DAC driver.
*/
#ifndef __LINUX_SPI_AT73C213_H
#define __LINUX_SPI_AT73C213_H
/**
* at73c213_board_info - how the external DAC is wired to the device.
*
* @ssc_id: SSC platform_driver id the DAC shall use to stream the audio.
* @dac_clk: the external clock used to provide master clock to the DAC.
* @shortname: a short discription for the DAC, seen by userspace tools.
*
* This struct contains the configuration of the hardware connection to the
* external DAC. The DAC needs a master clock and a I2S audio stream. It also
* provides a name which is used to identify it in userspace tools.
*/
struct at73c213_board_info {
int ssc_id;
struct clk *dac_clk;
char shortname[32];
};
#endif /* __LINUX_SPI_AT73C213_H */

8
include/sound/ac97_codec.h
View File

@ -2,7 +2,7 @@
#define __SOUND_AC97_CODEC_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.1
@ -345,9 +345,9 @@
#define AC97_ALC650_GPIO_STATUS 0x78
#define AC97_ALC650_CLOCK 0x7a
/* specific - Yamaha YMF753 */
#define AC97_YMF753_DIT_CTRL2 0x66 /* DIT Control 2 */
#define AC97_YMF753_3D_MODE_SEL 0x68 /* 3D Mode Select */
/* specific - Yamaha YMF7x3 */
#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */
#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */
/* specific - C-Media */
#define AC97_CM9738_VENDOR_CTRL 0x5a

5
include/sound/ad1848.h
View File

@ -2,7 +2,7 @@
#define __SOUND_AD1848_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Definitions for AD1847/AD1848/CS4248 chips
*
*
@ -27,7 +27,7 @@
/* IO ports */
#define AD1848P( codec, x ) ( (chip) -> port + c_d_c_AD1848##x )
#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x )
#define c_d_c_AD1848REGSEL 0
#define c_d_c_AD1848REG 1
@ -154,7 +154,6 @@ struct snd_ad1848 {
#endif
spinlock_t reg_lock;
struct mutex open_mutex;
};
/* exported functions */

2
include/sound/ainstr_gf1.h
View File

@ -2,7 +2,7 @@
* Advanced Linux Sound Architecture
*
* GF1 (GUS) Patch Instrument Format
* Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/ainstr_iw.h
View File

@ -2,7 +2,7 @@
* Advanced Linux Sound Architecture
*
* InterWave FFFF Instrument Format
* Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/ainstr_simple.h
View File

@ -2,7 +2,7 @@
* Advanced Linux Sound Architecture
*
* Simple (MOD player) Instrument Format
* Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/ak4114.h
View File

@ -3,7 +3,7 @@
/*
* Routines for Asahi Kasei AK4114
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/ak4117.h
View File

@ -3,7 +3,7 @@
/*
* Routines for Asahi Kasei AK4117
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/ak4531_codec.h
View File

@ -2,7 +2,7 @@
#define __SOUND_AK4531_CODEC_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.1

2
include/sound/ak4xxx-adda.h
View File

@ -5,7 +5,7 @@
* ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4381
* AD and DA converters
*
* Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

2
include/sound/asequencer.h
View File

@ -1,7 +1,7 @@
/*
* Main header file for the ALSA sequencer
* Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
* (c) 1998-1999 by Jaroslav Kysela <perex@suse.cz>
* (c) 1998-1999 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

3
include/sound/asound.h
View File

@ -1,6 +1,6 @@
/*
* Advanced Linux Sound Architecture - ALSA - Driver
* Copyright (c) 1994-2003 by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) 1994-2003 by Jaroslav Kysela <perex@perex.cz>,
* Abramo Bagnara <abramo@alsa-project.org>
*
*
@ -92,6 +92,7 @@ enum {
SNDRV_HWDEP_IFACE_USX2Y_PCM, /* Tascam US122, US224 & US428 rawusb pcm */
SNDRV_HWDEP_IFACE_PCXHR, /* Digigram PCXHR */
SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */
SNDRV_HWDEP_IFACE_HDA, /* HD-audio */
/* Don't forget to change the following: */
SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC

2
include/sound/asound_fm.h
View File

@ -5,7 +5,7 @@
* Advanced Linux Sound Architecture - ALSA
*
* Interface file between ALSA driver & user space
* Copyright (c) 1994-98 by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) 1994-98 by Jaroslav Kysela <perex@perex.cz>,
* 4Front Technologies
*
* Direct FM control

2
include/sound/asoundef.h
View File

@ -3,7 +3,7 @@
/*
* Advanced Linux Sound Architecture - ALSA - Driver
* Copyright (c) 1994-2000 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

10
include/sound/control.h
View File

@ -3,7 +3,7 @@
/*
* Header file for control interface
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -161,4 +161,12 @@ static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id
return dst_id;
}
/*
* Frequently used control callbacks
*/
int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
#endif /* __SOUND_CONTROL_H */

2
include/sound/core.h
View File

@ -3,7 +3,7 @@
/*
* Main header file for the ALSA driver
* Copyright (c) 1994-2001 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1994-2001 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

180
include/sound/cs4231-regs.h 100644
View File

@ -0,0 +1,180 @@
#ifndef __SOUND_CS4231_REGS_H
#define __SOUND_CS4231_REGS_H
/*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Definitions for CS4231 & InterWave chips & compatible chips registers
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/* IO ports */
#define CS4231P(x) (c_d_c_CS4231##x)
#define c_d_c_CS4231REGSEL 0
#define c_d_c_CS4231REG 1
#define c_d_c_CS4231STATUS 2
#define c_d_c_CS4231PIO 3
/* codec registers */
#define CS4231_LEFT_INPUT 0x00 /* left input control */
#define CS4231_RIGHT_INPUT 0x01 /* right input control */
#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */
#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */
#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */
#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */
#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */
#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */
#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */
#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */
#define CS4231_PIN_CTRL 0x0a /* pin control */
#define CS4231_TEST_INIT 0x0b /* test and initialization */
#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */
#define CS4231_LOOPBACK 0x0d /* loopback control */
#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */
#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */
#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */
#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */
#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */
#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */
#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */
#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */
#define CS4231_TIMER_LOW 0x14 /* timer low byte */
#define CS4231_TIMER_HIGH 0x15 /* timer high byte */
#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */
#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */
#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */
#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */
#define CS4236_EXT_REG 0x17 /* extended register access */
#define CS4231_IRQ_STATUS 0x18 /* irq status register */
#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */
#define CS4231_VERSION 0x19 /* CS4231(A) - version values */
#define CS4231_MONO_CTRL 0x1a /* mono input/output control */
#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */
#define AD1845_PWR_DOWN 0x1b /* power down control */
#define CS4235_LEFT_MASTER 0x1b /* left master output control */
#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */
#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */
#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */
#define CS4235_RIGHT_MASTER 0x1d /* right master output control */
#define CS4231_REC_UPR_CNT 0x1e /* record upper count */
#define CS4231_REC_LWR_CNT 0x1f /* record lower count */
/* definitions for codec register select port - CODECP( REGSEL ) */
#define CS4231_INIT 0x80 /* CODEC is initializing */
#define CS4231_MCE 0x40 /* mode change enable */
#define CS4231_TRD 0x20 /* transfer request disable */
/* definitions for codec status register - CODECP( STATUS ) */
#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */
/* definitions for codec irq status */
#define CS4231_PLAYBACK_IRQ 0x10
#define CS4231_RECORD_IRQ 0x20
#define CS4231_TIMER_IRQ 0x40
#define CS4231_ALL_IRQS 0x70
#define CS4231_REC_UNDERRUN 0x08
#define CS4231_REC_OVERRUN 0x04
#define CS4231_PLY_OVERRUN 0x02
#define CS4231_PLY_UNDERRUN 0x01
/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
#define CS4231_ENABLE_MIC_GAIN 0x20
#define CS4231_MIXS_LINE 0x00
#define CS4231_MIXS_AUX1 0x40
#define CS4231_MIXS_MIC 0x80
#define CS4231_MIXS_ALL 0xc0
/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */
#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */
#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */
#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */
#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */
#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */
#define CS4231_STEREO 0x10 /* stereo mode */
/* bits 3-1 define frequency divisor */
#define CS4231_XTAL1 0x00 /* 24.576 crystal */
#define CS4231_XTAL2 0x01 /* 16.9344 crystal */
/* definitions for interface control register - CS4231_IFACE_CTRL */
#define CS4231_RECORD_PIO 0x80 /* record PIO enable */
#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */
#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */
#define CS4231_AUTOCALIB 0x08 /* auto calibrate */
#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */
#define CS4231_RECORD_ENABLE 0x02 /* record enable */
#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */
/* definitions for pin control register - CS4231_PIN_CTRL */
#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */
#define CS4231_XCTL1 0x40 /* external control #1 */
#define CS4231_XCTL0 0x80 /* external control #0 */
/* definitions for test and init register - CS4231_TEST_INIT */
#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */
#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */
/* definitions for misc control register - CS4231_MISC_INFO */
#define CS4231_MODE2 0x40 /* MODE 2 */
#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */
#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */
/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
#define CS4231_DACZ 0x01 /* zero DAC when underrun */
#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */
#define CS4231_OLB 0x80 /* output level bit */
/* definitions for Extended Registers - CS4236+ */
#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */
#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */
#define CS4236_LEFT_MIC 0x28 /* left MIC volume */
#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */
#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */
#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */
#define CS4236_LEFT_FM 0x68 /* left FM volume */
#define CS4236_RIGHT_FM 0x78 /* right FM volume */
#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */
#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */
#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */
#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */
#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */
#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */
#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */
#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */
#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */
#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */
#define CS4236_VERSION 0x9c /* chip version and ID */
#endif /* __SOUND_CS4231_REGS_H */

159
include/sound/cs4231.h
View File

@ -2,7 +2,7 @@
#define __SOUND_CS4231_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Definitions for CS4231 & InterWave chips & compatible chips
*
*
@ -26,160 +26,7 @@
#include "pcm.h"
#include "timer.h"
/* IO ports */
#define CS4231P(x) (c_d_c_CS4231##x)
#define c_d_c_CS4231REGSEL 0
#define c_d_c_CS4231REG 1
#define c_d_c_CS4231STATUS 2
#define c_d_c_CS4231PIO 3
/* codec registers */
#define CS4231_LEFT_INPUT 0x00 /* left input control */
#define CS4231_RIGHT_INPUT 0x01 /* right input control */
#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */
#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */
#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */
#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */
#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */
#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */
#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */
#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */
#define CS4231_PIN_CTRL 0x0a /* pin control */
#define CS4231_TEST_INIT 0x0b /* test and initialization */
#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */
#define CS4231_LOOPBACK 0x0d /* loopback control */
#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */
#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */
#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */
#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */
#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */
#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */
#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */
#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */
#define CS4231_TIMER_LOW 0x14 /* timer low byte */
#define CS4231_TIMER_HIGH 0x15 /* timer high byte */
#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */
#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */
#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */
#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */
#define CS4236_EXT_REG 0x17 /* extended register access */
#define CS4231_IRQ_STATUS 0x18 /* irq status register */
#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */
#define CS4231_VERSION 0x19 /* CS4231(A) - version values */
#define CS4231_MONO_CTRL 0x1a /* mono input/output control */
#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */
#define AD1845_PWR_DOWN 0x1b /* power down control */
#define CS4235_LEFT_MASTER 0x1b /* left master output control */
#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */
#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */
#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */
#define CS4235_RIGHT_MASTER 0x1d /* right master output control */
#define CS4231_REC_UPR_CNT 0x1e /* record upper count */
#define CS4231_REC_LWR_CNT 0x1f /* record lower count */
/* definitions for codec register select port - CODECP( REGSEL ) */
#define CS4231_INIT 0x80 /* CODEC is initializing */
#define CS4231_MCE 0x40 /* mode change enable */
#define CS4231_TRD 0x20 /* transfer request disable */
/* definitions for codec status register - CODECP( STATUS ) */
#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */
/* definitions for codec irq status */
#define CS4231_PLAYBACK_IRQ 0x10
#define CS4231_RECORD_IRQ 0x20
#define CS4231_TIMER_IRQ 0x40
#define CS4231_ALL_IRQS 0x70
#define CS4231_REC_UNDERRUN 0x08
#define CS4231_REC_OVERRUN 0x04
#define CS4231_PLY_OVERRUN 0x02
#define CS4231_PLY_UNDERRUN 0x01
/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
#define CS4231_ENABLE_MIC_GAIN 0x20
#define CS4231_MIXS_LINE 0x00
#define CS4231_MIXS_AUX1 0x40
#define CS4231_MIXS_MIC 0x80
#define CS4231_MIXS_ALL 0xc0
/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */
#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */
#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */
#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */
#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */
#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */
#define CS4231_STEREO 0x10 /* stereo mode */
/* bits 3-1 define frequency divisor */
#define CS4231_XTAL1 0x00 /* 24.576 crystal */
#define CS4231_XTAL2 0x01 /* 16.9344 crystal */
/* definitions for interface control register - CS4231_IFACE_CTRL */
#define CS4231_RECORD_PIO 0x80 /* record PIO enable */
#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */
#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */
#define CS4231_AUTOCALIB 0x08 /* auto calibrate */
#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */
#define CS4231_RECORD_ENABLE 0x02 /* record enable */
#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */
/* definitions for pin control register - CS4231_PIN_CTRL */
#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */
#define CS4231_XCTL1 0x40 /* external control #1 */
#define CS4231_XCTL0 0x80 /* external control #0 */
/* definitions for test and init register - CS4231_TEST_INIT */
#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */
#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */
/* definitions for misc control register - CS4231_MISC_INFO */
#define CS4231_MODE2 0x40 /* MODE 2 */
#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */
#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */
/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
#define CS4231_DACZ 0x01 /* zero DAC when underrun */
#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */
#define CS4231_OLB 0x80 /* output level bit */
/* definitions for Extended Registers - CS4236+ */
#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */
#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */
#define CS4236_LEFT_MIC 0x28 /* left MIC volume */
#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */
#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */
#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */
#define CS4236_LEFT_FM 0x68 /* left FM volume */
#define CS4236_RIGHT_FM 0x78 /* right FM volume */
#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */
#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */
#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */
#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */
#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */
#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */
#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */
#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */
#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */
#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */
#define CS4236_VERSION 0x9c /* chip version and ID */
#include "cs4231-regs.h"
/* defines for codec.mode */
@ -210,7 +57,7 @@
#define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */
/* compatible, but clones */
#define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */
#define CS4231_HW_OPL3SA2 0x1001 /* OPL3-SA2 chip */
#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */
/* defines for codec.hwshare */
#define CS4231_HWSHARE_IRQ (1<<0)

2
include/sound/cs46xx.h
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@ -2,7 +2,7 @@
#define __SOUND_CS46XX_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Cirrus Logic, Inc.
* Definitions for Cirrus Logic CS46xx chips
*

2
include/sound/cs46xx_dsp_scb_types.h
View File

@ -1,6 +1,6 @@
/*
* The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/cs46xx_dsp_spos.h
View File

@ -1,6 +1,6 @@
/*
* The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/cs46xx_dsp_task_types.h
View File

@ -1,6 +1,6 @@
/*
* The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/cs8403.h
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@ -3,7 +3,7 @@
/*
* Routines for Cirrus Logic CS8403/CS8404A IEC958 (S/PDIF) Transmitter
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Takashi Iwai <tiwai@suse.de>
*
*

2
include/sound/cs8427.h
View File

@ -3,7 +3,7 @@
/*
* Routines for Cirrus Logic CS8427
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/driver.h
View File

@ -3,7 +3,7 @@
/*
* Main header file for the ALSA driver
* Copyright (c) 1994-2000 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

15
include/sound/emu10k1.h
View File

@ -2,7 +2,7 @@
#define __SOUND_EMU10K1_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Creative Labs, Inc.
* Definitions for EMU10K1 (SB Live!) chips
*
@ -1408,8 +1408,6 @@ struct snd_emu10k1_fx8010 {
struct snd_emu10k1_fx8010_irq *irq_handlers;
};
#define emu10k1_gpr_ctl(n) list_entry(n, struct snd_emu10k1_fx8010_ctl, list)
struct snd_emu10k1_midi {
struct snd_emu10k1 *emu;
struct snd_rawmidi *rmidi;
@ -1456,6 +1454,9 @@ struct snd_emu1010 {
unsigned int adc_pads; /* bit mask */
unsigned int dac_pads; /* bit mask */
unsigned int internal_clock; /* 44100 or 48000 */
unsigned int optical_in; /* 0:SPDIF, 1:ADAT */
unsigned int optical_out; /* 0:SPDIF, 1:ADAT */
struct task_struct *firmware_thread;
};
struct snd_emu10k1 {
@ -1599,9 +1600,9 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg,
void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data);
int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data);
int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value);
int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value);
int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value);
int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src);
int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value);
int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value);
int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src);
unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc);
void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb);
void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb);
@ -1746,6 +1747,8 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
#define A_FXBUS2(x) (0x80 + (x)) /* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */
#define A_EMU32OUTH(x) (0xa0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" - ??? */
#define A_EMU32OUTL(x) (0xb0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_1 - _F" - ??? */
#define A3_EMU32IN(x) (0x160 + (x)) /* x = 0x00 - 0x3f "EMU32_IN_00 - _3F" - Only when .device = 0x0008 */
#define A3_EMU32OUT(x) (0x1E0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_00 - _3F" - Only when .device = 0x0008 */
#define A_GPR(x) (A_FXGPREGBASE + (x))
/* cc_reg constants */

2
include/sound/es1688.h
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@ -3,7 +3,7 @@
/*
* Header file for ES488/ES1688
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/gus.h
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@ -3,7 +3,7 @@
/*
* Global structures used for GUS part of ALSA driver
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

44
include/sound/hda_hwdep.h 100644
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@ -0,0 +1,44 @@
/*
* HWDEP Interface for HD-audio codec
*
* Copyright (c) 2007 Takashi Iwai <tiwai@suse.de>
*
* This driver is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This driver is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#ifndef __SOUND_HDA_HWDEP_H
#define __SOUND_HDA_HWDEP_H
#define HDA_HWDEP_VERSION ((1 << 16) | (0 << 8) | (0 << 0)) /* 1.0.0 */
/* verb */
#define HDA_REG_NID_SHIFT 24
#define HDA_REG_VERB_SHIFT 8
#define HDA_REG_VAL_SHIFT 0
#define HDA_VERB(nid,verb,param) ((nid)<<24 | (verb)<<8 | (param))
struct hda_verb_ioctl {
u32 verb; /* HDA_VERB() */
u32 res; /* response */
};
/*
* ioctls
*/
#define HDA_IOCTL_PVERSION _IOR('H', 0x10, int)
#define HDA_IOCTL_VERB_WRITE _IOWR('H', 0x11, struct hda_verb_ioctl)
#define HDA_IOCTL_GET_WCAP _IOWR('H', 0x12, struct hda_verb_ioctl)
#endif

16
include/sound/hdspm.h
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@ -1,4 +1,4 @@
#ifndef __SOUND_HDSPM_H /* -*- linux-c -*- */
#ifndef __SOUND_HDSPM_H
#define __SOUND_HDSPM_H
/*
* Copyright (C) 2003 Winfried Ritsch (IEM)
@ -61,7 +61,8 @@ struct hdspm_peak_rms_ioctl {
};
/* use indirect access due to the limit of ioctl bit size */
#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct hdspm_peak_rms_ioctl)
#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \
_IOR('H', 0x40, struct hdspm_peak_rms_ioctl)
/* ------------ CONFIG block IOCTL ---------------------- */
@ -79,7 +80,8 @@ struct hdspm_config_info {
unsigned int analog_out;
};
#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, struct hdspm_config_info)
#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \
_IOR('H', 0x41, struct hdspm_config_info)
/* get Soundcard Version */
@ -93,10 +95,14 @@ struct hdspm_version {
/* ------------- get Matrix Mixer IOCTL --------------- */
/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = 32768 Bytes */
/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte =
* 32768 Bytes
*/
/* organisation is 64 channelfader in a continous memory block */
/* equivalent to hardware definition, maybe for future feature of mmap of them */
/* equivalent to hardware definition, maybe for future feature of mmap of
* them
*/
/* each of 64 outputs has 64 infader and 64 outfader:
Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */

2
include/sound/hwdep.h
View File

@ -3,7 +3,7 @@
/*
* Hardware dependent layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/info.h
View File

@ -3,7 +3,7 @@
/*
* Header file for info interface
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/initval.h
View File

@ -3,7 +3,7 @@
/*
* Init values for soundcard modules
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

2
include/sound/memalloc.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Takashi Iwai <tiwai@suse.de>
*
* Generic memory allocators

2
include/sound/mixer_oss.h
View File

@ -3,7 +3,7 @@
/*
* OSS MIXER API
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

3
include/sound/mpu401.h
View File

@ -3,7 +3,7 @@
/*
* Header file for MPU-401 and compatible cards
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -50,7 +50,6 @@
#define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */
#define MPU401_INFO_MMIO (1 << 3) /* MMIO access */
#define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */
#define MPU401_INFO_UART_ONLY (1 << 5) /* No ENTER_UART cmd needed */
#define MPU401_MODE_BIT_INPUT 0
#define MPU401_MODE_BIT_OUTPUT 1

2
include/sound/opl3.h
View File

@ -4,7 +4,7 @@
/*
* Definitions of the OPL-3 registers.
*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Hannu Savolainen 1993-1996
*
*

2
include/sound/pcm-indirect.h
View File

@ -2,7 +2,7 @@
* Helper functions for indirect PCM data transfer
*
* Copyright (c) by Takashi Iwai <tiwai@suse.de>
* Jaroslav Kysela <perex@suse.cz>
* Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

13
include/sound/pcm.h
View File

@ -3,7 +3,7 @@
/*
* Digital Audio (PCM) abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Abramo Bagnara <abramo@alsa-project.org>
*
*
@ -301,8 +301,8 @@ struct snd_pcm_runtime {
union snd_pcm_sync_id sync; /* hardware synchronization ID */
/* -- mmap -- */
volatile struct snd_pcm_mmap_status *status;
volatile struct snd_pcm_mmap_control *control;
struct snd_pcm_mmap_status *status;
struct snd_pcm_mmap_control *control;
/* -- locking / scheduling -- */
wait_queue_head_t sleep;
@ -791,13 +791,13 @@ static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *p
static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var)
{
return (const struct snd_mask *)hw_param_mask((struct snd_pcm_hw_params*) params, var);
return &params->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK];
}
static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var)
{
return (const struct snd_interval *)hw_param_interval((struct snd_pcm_hw_params*) params, var);
return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL];
}
#define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS))
@ -922,7 +922,10 @@ snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream,
snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream,
void __user **bufs, snd_pcm_uframes_t frames);
extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates;
int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream,
struct snd_dma_buffer *bufp)

2
include/sound/pcm_oss.h
View File

@ -3,7 +3,7 @@
/*
* Digital Audio (PCM) - OSS compatibility abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/rawmidi.h
View File

@ -3,7 +3,7 @@
/*
* Abstract layer for MIDI v1.0 stream
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/sb.h
View File

@ -3,7 +3,7 @@
/*
* Header file for SoundBlaster cards
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/seq_instr.h
View File

@ -3,7 +3,7 @@
/*
* Main kernel header file for the ALSA sequencer
* Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
include/sound/seq_midi_event.h
View File

@ -5,7 +5,7 @@
* MIDI byte <-> sequencer event coder
*
* Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>,
* Jaroslav Kysela <perex@suse.cz>
* Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

2
include/sound/seq_virmidi.h
View File

@ -4,7 +4,7 @@
/*
* Virtual Raw MIDI client on Sequencer
* Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>,
* Jaroslav Kysela <perex@suse.cz>
* Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

3
include/sound/soc.h
View File

@ -201,8 +201,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
#define snd_soc_info_bool_ext snd_ctl_boolean_mono
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,

2
include/sound/tea575x-tuner.h
View File

@ -4,7 +4,7 @@
/*
* ALSA driver for TEA5757/5759 Philips AM/FM tuner chips
*
* Copyright (c) 2004 Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by

2
include/sound/timer.h
View File

@ -3,7 +3,7 @@
/*
* Timer abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Abramo Bagnara <abramo@alsa-project.org>
*
*

2
include/sound/tlv.h
View File

@ -3,7 +3,7 @@
/*
* Advanced Linux Sound Architecture - ALSA - Driver
* Copyright (c) 2006 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 2006 by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

4
include/sound/version.h
View File

@ -1,3 +1,3 @@
/* include/version.h. Generated by alsa/ksync script. */
#define CONFIG_SND_VERSION "1.0.14"
#define CONFIG_SND_DATE " (Fri Jul 20 09:12:58 2007 UTC)"
#define CONFIG_SND_VERSION "1.0.15"
#define CONFIG_SND_DATE " (Tue Oct 16 14:57:44 2007 UTC)"

2
include/sound/ymfpci.h
View File

@ -2,7 +2,7 @@
#define __SOUND_YMFPCI_H
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Definitions for Yahama YMF724/740/744/754 chips
*
*

4
sound/Kconfig
View File

@ -63,6 +63,10 @@ source "sound/aoa/Kconfig"
source "sound/arm/Kconfig"
if SPI
source "sound/spi/Kconfig"
endif
source "sound/mips/Kconfig"
source "sound/sh/Kconfig"

3
sound/Makefile
View File

@ -5,7 +5,8 @@ obj-$(CONFIG_SOUND) += soundcore.o
obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
obj-$(CONFIG_SOUND_PRIME) += oss/
obj-$(CONFIG_DMASOUND) += oss/
obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
sparc/ spi/ parisc/ pcmcia/ mips/ soc/
obj-$(CONFIG_SND_AOA) += aoa/
# This one must be compilable even if sound is configured out

20
sound/aoa/codecs/snd-aoa-codec-onyx.c
View File

@ -297,15 +297,7 @@ static struct snd_kcontrol_new capture_source_control = {
.put = onyx_snd_capture_source_put,
};
static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define onyx_snd_mute_info snd_ctl_boolean_stereo_info
static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@ -359,15 +351,7 @@ static struct snd_kcontrol_new mute_control = {
};
static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define onyx_snd_single_bit_info snd_ctl_boolean_mono_info
#define FLAG_POLARITY_INVERT 1
#define FLAG_SPDIFLOCK 2

29
sound/aoa/codecs/snd-aoa-codec-tas.c
View File

@ -272,15 +272,7 @@ static struct snd_kcontrol_new volume_control = {
.put = tas_snd_vol_put,
};
static int tas_snd_mute_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define tas_snd_mute_info snd_ctl_boolean_stereo_info
static int tas_snd_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@ -431,15 +423,7 @@ static struct snd_kcontrol_new drc_range_control = {
.put = tas_snd_drc_range_put,
};
static int tas_snd_drc_switch_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define tas_snd_drc_switch_info snd_ctl_boolean_mono_info
static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@ -743,6 +727,7 @@ static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock
return 0;
}
#ifdef CONFIG_PM
/* we are controlled via i2c and assume that is always up
* If that wasn't the case, we'd have to suspend once
* our i2c device is suspended, and then take note of that! */
@ -768,7 +753,6 @@ static int tas_resume(struct tas *tas)
return 0;
}
#ifdef CONFIG_PM
static int _tas_suspend(struct codec_info_item *cii, pm_message_t state)
{
return tas_suspend(cii->codec_data);
@ -778,7 +762,10 @@ static int _tas_resume(struct codec_info_item *cii)
{
return tas_resume(cii->codec_data);
}
#endif
#else /* CONFIG_PM */
#define _tas_suspend NULL
#define _tas_resume NULL
#endif /* CONFIG_PM */
static struct codec_info tas_codec_info = {
.transfers = tas_transfers,
@ -791,10 +778,8 @@ static struct codec_info tas_codec_info = {
.owner = THIS_MODULE,
.usable = tas_usable,
.switch_clock = tas_switch_clock,
#ifdef CONFIG_PM
.suspend = _tas_suspend,
.resume = _tas_resume,
#endif
};
static int tas_init_codec(struct aoa_codec *codec)

10
sound/aoa/fabrics/snd-aoa-fabric-layout.c
View File

@ -582,15 +582,7 @@ static int layouts_list_items;
* make the fabric handle all the card stuff, etc... */
static struct layout_dev *layout_device;
static int control_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define control_info snd_ctl_boolean_mono_info
#define AMP_CONTROL(n, description) \
static int n##_control_get(struct snd_kcontrol *kcontrol, \

35
sound/arm/sa11xx-uda1341.c
View File

@ -79,12 +79,6 @@
#include <asm/mach-types.h>
#include <asm/dma.h>
#ifdef CONFIG_H3600_HAL
#include <asm/semaphore.h>
#include <asm/uaccess.h>
#include <asm/arch/h3600_hal.h>
#endif
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
@ -100,9 +94,6 @@
* We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
* module for Familiar 0.6.1
*/
#ifdef CONFIG_H3600_HAL
#define HH_VERSION 1
#endif
/* {{{ Type definitions */
@ -238,11 +229,8 @@ static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341,
rate = 8000;
/* Set the external clock generator */
#ifdef CONFIG_H3600_HAL
h3600_audio_clock(rate);
#else
sa11xx_uda1341_set_audio_clock(rate);
#endif
/* Select the clock divisor */
switch (rate) {
@ -307,13 +295,10 @@ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
local_irq_restore(flags);
/* Enable the audio power */
#ifdef CONFIG_H3600_HAL
h3600_audio_power(AUDIO_RATE_DEFAULT);
#else
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
/* Wait for the UDA1341 to wake up */
mdelay(1); //FIXME - was removed by Perex - Why?
@ -331,21 +316,13 @@ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
/* make the left and right channels unswapped (flip the WS latch) */
Ser4SSDR = 0;
#ifdef CONFIG_H3600_HAL
h3600_audio_mute(0);
#else
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
}
static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
{
/* mute on */
#ifdef CONFIG_H3600_HAL
h3600_audio_mute(1);
#else
set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
/* disable the audio power and all signals leading to the audio chip */
l3_close(sa11xx_uda1341->uda1341);
@ -354,13 +331,9 @@ static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
/* power off and mute off */
/* FIXME - is muting off necesary??? */
#ifdef CONFIG_H3600_HAL
h3600_audio_power(0);
h3600_audio_mute(0);
#else
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
#endif
}
/* }}} */

15
sound/core/Makefile
View File

@ -1,20 +1,17 @@
#
# Makefile for ALSA
# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@suse.cz>
# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-objs := sound.o init.o memory.o info.o control.o misc.o device.o
ifeq ($(CONFIG_ISA_DMA_API),y)
snd-objs += isadma.o
endif
ifeq ($(CONFIG_SND_OSSEMUL),y)
snd-objs += sound_oss.o info_oss.o
endif
snd-y := sound.o init.o memory.o info.o control.o misc.o device.o
snd-$(CONFIG_ISA_DMA_API) += isadma.o
snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o
snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
pcm_memory.o
snd-page-alloc-objs := memalloc.o sgbuf.o
snd-page-alloc-y := memalloc.o
snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o
snd-rawmidi-objs := rawmidi.o
snd-timer-objs := timer.o

33
sound/core/control.c
View File

@ -1,6 +1,6 @@
/*
* Routines for driver control interface
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -716,8 +716,6 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control)
return result;
}
EXPORT_SYMBOL(snd_ctl_elem_read);
static int snd_ctl_elem_read_user(struct snd_card *card,
struct snd_ctl_elem_value __user *_control)
{
@ -781,8 +779,6 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
return result;
}
EXPORT_SYMBOL(snd_ctl_elem_write);
static int snd_ctl_elem_write_user(struct snd_ctl_file *file,
struct snd_ctl_elem_value __user *_control)
{
@ -1486,3 +1482,30 @@ int snd_ctl_create(struct snd_card *card)
snd_assert(card != NULL, return -ENXIO);
return snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops);
}
/*
* Frequently used control callbacks
*/
int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
EXPORT_SYMBOL(snd_ctl_boolean_mono_info);
int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
EXPORT_SYMBOL(snd_ctl_boolean_stereo_info);

2
sound/core/device.c
View File

@ -1,6 +1,6 @@
/*
* Device management routines
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

4
sound/core/hwdep.c
View File

@ -1,6 +1,6 @@
/*
* Hardware dependent layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -31,7 +31,7 @@
#include <sound/hwdep.h>
#include <sound/info.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Hardware dependent layer");
MODULE_LICENSE("GPL");

2
sound/core/info.c
View File

@ -1,6 +1,6 @@
/*
* Information interface for ALSA driver
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
sound/core/info_oss.c
View File

@ -1,6 +1,6 @@
/*
* Information interface for ALSA driver
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
sound/core/init.c
View File

@ -1,6 +1,6 @@
/*
* Initialization routines
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

2
sound/core/isadma.c
View File

@ -1,6 +1,6 @@
/*
* ISA DMA support functions
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

10
sound/core/memalloc.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Takashi Iwai <tiwai@suse.de>
*
* Generic memory allocators
@ -38,7 +38,7 @@
#endif
MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@suse.cz>");
MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Memory allocator for ALSA system.");
MODULE_LICENSE("GPL");
@ -206,6 +206,7 @@ void snd_free_pages(void *ptr, size_t size)
*
*/
#ifdef CONFIG_HAS_DMA
/* allocate the coherent DMA pages */
static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
{
@ -239,6 +240,7 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
dec_snd_pages(pg);
dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
}
#endif /* CONFIG_HAS_DMA */
#ifdef CONFIG_SBUS
@ -312,12 +314,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr);
break;
#endif
#ifdef CONFIG_HAS_DMA
case SNDRV_DMA_TYPE_DEV:
dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
break;
case SNDRV_DMA_TYPE_DEV_SG:
snd_malloc_sgbuf_pages(device, size, dmab, NULL);
break;
#endif
default:
printk(KERN_ERR "snd-malloc: invalid device type %d\n", type);
dmab->area = NULL;
@ -383,12 +387,14 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
break;
#endif
#ifdef CONFIG_HAS_DMA
case SNDRV_DMA_TYPE_DEV:
snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
break;
case SNDRV_DMA_TYPE_DEV_SG:
snd_free_sgbuf_pages(dmab);
break;
#endif
default:
printk(KERN_ERR "snd-malloc: invalid device type %d\n", dmab->dev.type);
}

2
sound/core/memory.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* Misc memory accessors
*

2
sound/core/misc.c
View File

@ -1,6 +1,6 @@
/*
* Misc and compatibility things
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

7
sound/core/oss/Makefile
View File

@ -1,12 +1,13 @@
#
# Makefile for ALSA
# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
snd-mixer-oss-objs := mixer_oss.o
snd-pcm-oss-objs := pcm_oss.o pcm_plugin.o \
io.o copy.o linear.o mulaw.o route.o rate.o
snd-pcm-oss-y := pcm_oss.o
snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \
io.o copy.o linear.o mulaw.o route.o rate.o
obj-$(CONFIG_SND_MIXER_OSS) += snd-mixer-oss.o
obj-$(CONFIG_SND_PCM_OSS) += snd-pcm-oss.o

5
sound/core/oss/copy.c
View File

@ -20,9 +20,6 @@
*/
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -88,5 +85,3 @@ int snd_pcm_plugin_build_copy(struct snd_pcm_substream *plug,
*r_plugin = plugin;
return 0;
}
#endif

7
sound/core/oss/io.c
View File

@ -1,6 +1,6 @@
/*
* PCM I/O Plug-In Interface
* Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
*
*
* This library is free software; you can redistribute it and/or modify
@ -20,9 +20,6 @@
*/
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -135,5 +132,3 @@ int snd_pcm_plugin_build_io(struct snd_pcm_substream *plug,
*r_plugin = plugin;
return 0;
}
#endif

87
sound/core/oss/linear.c
View File

@ -1,6 +1,6 @@
/*
* Linear conversion Plug-In
* Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>,
* Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>,
* Abramo Bagnara <abramo@alsa-project.org>
*
*
@ -21,9 +21,6 @@
*/
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -34,19 +31,34 @@
*/
struct linear_priv {
int conv;
int cvt_endian; /* need endian conversion? */
unsigned int src_ofs; /* byte offset in source format */
unsigned int dst_ofs; /* byte soffset in destination format */
unsigned int copy_ofs; /* byte offset in temporary u32 data */
unsigned int dst_bytes; /* byte size of destination format */
unsigned int copy_bytes; /* bytes to copy per conversion */
unsigned int flip; /* MSB flip for signeness, done after endian conv */
};
static inline void do_convert(struct linear_priv *data,
unsigned char *dst, unsigned char *src)
{
unsigned int tmp = 0;
unsigned char *p = (unsigned char *)&tmp;
memcpy(p + data->copy_ofs, src + data->src_ofs, data->copy_bytes);
if (data->cvt_endian)
tmp = swab32(tmp);
tmp ^= data->flip;
memcpy(dst, p + data->dst_ofs, data->dst_bytes);
}
static void convert(struct snd_pcm_plugin *plugin,
const struct snd_pcm_plugin_channel *src_channels,
struct snd_pcm_plugin_channel *dst_channels,
snd_pcm_uframes_t frames)
{
#define CONV_LABELS
#include "plugin_ops.h"
#undef CONV_LABELS
struct linear_priv *data = (struct linear_priv *)plugin->extra_data;
void *conv = conv_labels[data->conv];
int channel;
int nchannels = plugin->src_format.channels;
for (channel = 0; channel < nchannels; ++channel) {
@ -67,11 +79,7 @@ static void convert(struct snd_pcm_plugin *plugin,
dst_step = dst_channels[channel].area.step / 8;
frames1 = frames;
while (frames1-- > 0) {
goto *conv;
#define CONV_END after
#include "plugin_ops.h"
#undef CONV_END
after:
do_convert(data, dst, src);
src += src_step;
dst += dst_step;
}
@ -106,29 +114,36 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin,
return frames;
}
static int conv_index(int src_format, int dst_format)
static void init_data(struct linear_priv *data, int src_format, int dst_format)
{
int src_endian, dst_endian, sign, src_width, dst_width;
int src_le, dst_le, src_bytes, dst_bytes;
sign = (snd_pcm_format_signed(src_format) !=
snd_pcm_format_signed(dst_format));
#ifdef SNDRV_LITTLE_ENDIAN
src_endian = snd_pcm_format_big_endian(src_format);
dst_endian = snd_pcm_format_big_endian(dst_format);
#else
src_endian = snd_pcm_format_little_endian(src_format);
dst_endian = snd_pcm_format_little_endian(dst_format);
#endif
src_bytes = snd_pcm_format_width(src_format) / 8;
dst_bytes = snd_pcm_format_width(dst_format) / 8;
src_le = snd_pcm_format_little_endian(src_format) > 0;
dst_le = snd_pcm_format_little_endian(dst_format) > 0;
if (src_endian < 0)
src_endian = 0;
if (dst_endian < 0)
dst_endian = 0;
src_width = snd_pcm_format_width(src_format) / 8 - 1;
dst_width = snd_pcm_format_width(dst_format) / 8 - 1;
return src_width * 32 + src_endian * 16 + sign * 8 + dst_width * 2 + dst_endian;
data->dst_bytes = dst_bytes;
data->cvt_endian = src_le != dst_le;
data->copy_bytes = src_bytes < dst_bytes ? src_bytes : dst_bytes;
if (src_le) {
data->copy_ofs = 4 - data->copy_bytes;
data->src_ofs = src_bytes - data->copy_bytes;
} else
data->src_ofs = snd_pcm_format_physical_width(src_format) / 8 -
src_bytes;
if (dst_le)
data->dst_ofs = 4 - data->dst_bytes;
else
data->dst_ofs = snd_pcm_format_physical_width(dst_format) / 8 -
dst_bytes;
if (snd_pcm_format_signed(src_format) !=
snd_pcm_format_signed(dst_format)) {
if (dst_le)
data->flip = cpu_to_le32(0x80000000);
else
data->flip = cpu_to_be32(0x80000000);
}
}
int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug,
@ -154,10 +169,8 @@ int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug,
if (err < 0)
return err;
data = (struct linear_priv *)plugin->extra_data;
data->conv = conv_index(src_format->format, dst_format->format);
init_data(data, src_format->format, dst_format->format);
plugin->transfer = linear_transfer;
*r_plugin = plugin;
return 0;
}
#endif

4
sound/core/oss/mixer_oss.c
View File

@ -1,6 +1,6 @@
/*
* OSS emulation layer for the mixer interface
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -33,7 +33,7 @@
#define OSS_ALSAEMULVER _SIOR ('M', 249, int)
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Mixer OSS emulation for ALSA.");
MODULE_LICENSE("GPL");
MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MIXER);

90
sound/core/oss/mulaw.c
View File

@ -1,6 +1,6 @@
/*
* Mu-Law conversion Plug-In Interface
* Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
* Uros Bizjak <uros@kss-loka.si>
*
* Based on reference implementation by Sun Microsystems, Inc.
@ -22,9 +22,6 @@
*/
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -149,19 +146,32 @@ typedef void (*mulaw_f)(struct snd_pcm_plugin *plugin,
struct mulaw_priv {
mulaw_f func;
int conv;
int cvt_endian; /* need endian conversion? */
unsigned int native_ofs; /* byte offset in native format */
unsigned int copy_ofs; /* byte offset in s16 format */
unsigned int native_bytes; /* byte size of the native format */
unsigned int copy_bytes; /* bytes to copy per conversion */
u16 flip; /* MSB flip for signedness, done after endian conversion */
};
static inline void cvt_s16_to_native(struct mulaw_priv *data,
unsigned char *dst, u16 sample)
{
sample ^= data->flip;
if (data->cvt_endian)
sample = swab16(sample);
if (data->native_bytes > data->copy_bytes)
memset(dst, 0, data->native_bytes);
memcpy(dst + data->native_ofs, (char *)&sample + data->copy_ofs,
data->copy_bytes);
}
static void mulaw_decode(struct snd_pcm_plugin *plugin,
const struct snd_pcm_plugin_channel *src_channels,
struct snd_pcm_plugin_channel *dst_channels,
snd_pcm_uframes_t frames)
{
#define PUT_S16_LABELS
#include "plugin_ops.h"
#undef PUT_S16_LABELS
struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data;
void *put = put_s16_labels[data->conv];
int channel;
int nchannels = plugin->src_format.channels;
for (channel = 0; channel < nchannels; ++channel) {
@ -183,30 +193,33 @@ static void mulaw_decode(struct snd_pcm_plugin *plugin,
frames1 = frames;
while (frames1-- > 0) {
signed short sample = ulaw2linear(*src);
goto *put;
#define PUT_S16_END after
#include "plugin_ops.h"
#undef PUT_S16_END
after:
cvt_s16_to_native(data, dst, sample);
src += src_step;
dst += dst_step;
}
}
}
static inline signed short cvt_native_to_s16(struct mulaw_priv *data,
unsigned char *src)
{
u16 sample = 0;
memcpy((char *)&sample + data->copy_ofs, src + data->native_ofs,
data->copy_bytes);
if (data->cvt_endian)
sample = swab16(sample);
sample ^= data->flip;
return (signed short)sample;
}
static void mulaw_encode(struct snd_pcm_plugin *plugin,
const struct snd_pcm_plugin_channel *src_channels,
struct snd_pcm_plugin_channel *dst_channels,
snd_pcm_uframes_t frames)
{
#define GET_S16_LABELS
#include "plugin_ops.h"
#undef GET_S16_LABELS
struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data;
void *get = get_s16_labels[data->conv];
int channel;
int nchannels = plugin->src_format.channels;
signed short sample = 0;
for (channel = 0; channel < nchannels; ++channel) {
char *src;
char *dst;
@ -225,11 +238,7 @@ static void mulaw_encode(struct snd_pcm_plugin *plugin,
dst_step = dst_channels[channel].area.step / 8;
frames1 = frames;
while (frames1-- > 0) {
goto *get;
#define GET_S16_END after
#include "plugin_ops.h"
#undef GET_S16_END
after:
signed short sample = cvt_native_to_s16(data, src);
*dst = linear2ulaw(sample);
src += src_step;
dst += dst_step;
@ -265,23 +274,25 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin,
return frames;
}
static int getput_index(int format)
static void init_data(struct mulaw_priv *data, int format)
{
int sign, width, endian;
sign = !snd_pcm_format_signed(format);
width = snd_pcm_format_width(format) / 8 - 1;
if (width < 0 || width > 3) {
snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format);
width = 0;
}
#ifdef SNDRV_LITTLE_ENDIAN
endian = snd_pcm_format_big_endian(format);
data->cvt_endian = snd_pcm_format_big_endian(format) > 0;
#else
endian = snd_pcm_format_little_endian(format);
data->cvt_endian = snd_pcm_format_little_endian(format) > 0;
#endif
if (endian < 0)
endian = 0;
return width * 4 + endian * 2 + sign;
if (!snd_pcm_format_signed(format))
data->flip = 0x8000;
data->native_bytes = snd_pcm_format_physical_width(format) / 8;
data->copy_bytes = data->native_bytes < 2 ? 1 : 2;
if (snd_pcm_format_little_endian(format)) {
data->native_ofs = data->native_bytes - data->copy_bytes;
data->copy_ofs = 2 - data->copy_bytes;
} else {
/* S24 in 4bytes need an 1 byte offset */
data->native_ofs = data->native_bytes -
snd_pcm_format_width(format) / 8;
}
}
int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
@ -322,11 +333,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
return err;
data = (struct mulaw_priv *)plugin->extra_data;
data->func = func;
data->conv = getput_index(format->format);
snd_assert(data->conv >= 0 && data->conv < 4*2*2, return -EINVAL);
init_data(data, format->format);
plugin->transfer = mulaw_transfer;
*r_plugin = plugin;
return 0;
}
#endif

39
sound/core/oss/pcm_oss.c
View File

@ -1,6 +1,6 @@
/*
* Digital Audio (PCM) abstract layer / OSS compatible
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -48,7 +48,7 @@ static int dsp_map[SNDRV_CARDS];
static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1};
static int nonblock_open = 1;
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Abramo Bagnara <abramo@alsa-project.org>");
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>");
MODULE_DESCRIPTION("PCM OSS emulation for ALSA.");
MODULE_LICENSE("GPL");
module_param_array(dsp_map, int, NULL, 0444);
@ -633,6 +633,22 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes)
return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes);
}
/* define extended formats in the recent OSS versions (if any) */
/* linear formats */
#define AFMT_S32_LE 0x00001000
#define AFMT_S32_BE 0x00002000
#define AFMT_S24_LE 0x00008000
#define AFMT_S24_BE 0x00010000
#define AFMT_S24_PACKED 0x00040000
/* other supported formats */
#define AFMT_FLOAT 0x00004000
#define AFMT_SPDIF_RAW 0x00020000
/* unsupported formats */
#define AFMT_AC3 0x00000400
#define AFMT_VORBIS 0x00000800
static int snd_pcm_oss_format_from(int format)
{
switch (format) {
@ -646,6 +662,13 @@ static int snd_pcm_oss_format_from(int format)
case AFMT_U16_LE: return SNDRV_PCM_FORMAT_U16_LE;
case AFMT_U16_BE: return SNDRV_PCM_FORMAT_U16_BE;
case AFMT_MPEG: return SNDRV_PCM_FORMAT_MPEG;
case AFMT_S32_LE: return SNDRV_PCM_FORMAT_S32_LE;
case AFMT_S32_BE: return SNDRV_PCM_FORMAT_S32_BE;
case AFMT_S24_LE: return SNDRV_PCM_FORMAT_S24_LE;
case AFMT_S24_BE: return SNDRV_PCM_FORMAT_S24_BE;
case AFMT_S24_PACKED: return SNDRV_PCM_FORMAT_S24_3LE;
case AFMT_FLOAT: return SNDRV_PCM_FORMAT_FLOAT;
case AFMT_SPDIF_RAW: return SNDRV_PCM_FORMAT_IEC958_SUBFRAME;
default: return SNDRV_PCM_FORMAT_U8;
}
}
@ -663,6 +686,13 @@ static int snd_pcm_oss_format_to(int format)
case SNDRV_PCM_FORMAT_U16_LE: return AFMT_U16_LE;
case SNDRV_PCM_FORMAT_U16_BE: return AFMT_U16_BE;
case SNDRV_PCM_FORMAT_MPEG: return AFMT_MPEG;
case SNDRV_PCM_FORMAT_S32_LE: return AFMT_S32_LE;
case SNDRV_PCM_FORMAT_S32_BE: return AFMT_S32_BE;
case SNDRV_PCM_FORMAT_S24_LE: return AFMT_S24_LE;
case SNDRV_PCM_FORMAT_S24_BE: return AFMT_S24_BE;
case SNDRV_PCM_FORMAT_S24_3LE: return AFMT_S24_PACKED;
case SNDRV_PCM_FORMAT_FLOAT: return AFMT_FLOAT;
case SNDRV_PCM_FORMAT_IEC958_SUBFRAME: return AFMT_SPDIF_RAW;
default: return -EINVAL;
}
}
@ -1725,7 +1755,10 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
return AFMT_MU_LAW | AFMT_U8 |
AFMT_S16_LE | AFMT_S16_BE |
AFMT_S8 | AFMT_U16_LE |
AFMT_U16_BE;
AFMT_U16_BE |
AFMT_S32_LE | AFMT_S32_BE |
AFMT_S24_LE | AFMT_S24_LE |
AFMT_S24_PACKED;
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;

63
sound/core/oss/pcm_plugin.c
View File

@ -1,6 +1,6 @@
/*
* PCM Plug-In shared (kernel/library) code
* Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
* Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
*
*
@ -25,9 +25,6 @@
#endif
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/vmalloc.h>
@ -267,6 +264,8 @@ static int snd_pcm_plug_formats(struct snd_mask *mask, int format)
SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE |
SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE |
SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE |
SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE);
snd_mask_set(&formats, SNDRV_PCM_FORMAT_MU_LAW);
@ -283,6 +282,10 @@ static int preferred_formats[] = {
SNDRV_PCM_FORMAT_S16_BE,
SNDRV_PCM_FORMAT_U16_LE,
SNDRV_PCM_FORMAT_U16_BE,
SNDRV_PCM_FORMAT_S24_3LE,
SNDRV_PCM_FORMAT_S24_3BE,
SNDRV_PCM_FORMAT_U24_3LE,
SNDRV_PCM_FORMAT_U24_3BE,
SNDRV_PCM_FORMAT_S24_LE,
SNDRV_PCM_FORMAT_S24_BE,
SNDRV_PCM_FORMAT_U24_LE,
@ -297,41 +300,37 @@ static int preferred_formats[] = {
int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask)
{
int i;
if (snd_mask_test(format_mask, format))
return format;
if (! snd_pcm_plug_formats(format_mask, format))
return -EINVAL;
if (snd_pcm_format_linear(format)) {
int width = snd_pcm_format_width(format);
int unsignd = snd_pcm_format_unsigned(format);
int big = snd_pcm_format_big_endian(format);
int format1;
int wid, width1=width;
int dwidth1 = 8;
for (wid = 0; wid < 4; ++wid) {
int end, big1 = big;
for (end = 0; end < 2; ++end) {
int sgn, unsignd1 = unsignd;
for (sgn = 0; sgn < 2; ++sgn) {
format1 = snd_pcm_build_linear_format(width1, unsignd1, big1);
if (format1 >= 0 &&
snd_mask_test(format_mask, format1))
goto _found;
unsignd1 = !unsignd1;
}
big1 = !big1;
unsigned int width = snd_pcm_format_width(format);
int unsignd = snd_pcm_format_unsigned(format) > 0;
int big = snd_pcm_format_big_endian(format) > 0;
unsigned int badness, best = -1;
int best_format = -1;
for (i = 0; i < ARRAY_SIZE(preferred_formats); i++) {
int f = preferred_formats[i];
unsigned int w;
if (!snd_mask_test(format_mask, f))
continue;
w = snd_pcm_format_width(f);
if (w >= width)
badness = w - width;
else
badness = width - w + 32;
badness += snd_pcm_format_unsigned(f) != unsignd;
badness += snd_pcm_format_big_endian(f) != big;
if (badness < best) {
best_format = f;
best = badness;
}
if (width1 == 32) {
dwidth1 = -dwidth1;
width1 = width;
}
width1 += dwidth1;
}
return -EINVAL;
_found:
return format1;
return best_format >= 0 ? best_format : -EINVAL;
} else {
unsigned int i;
switch (format) {
case SNDRV_PCM_FORMAT_MU_LAW:
for (i = 0; i < ARRAY_SIZE(preferred_formats); ++i) {
@ -740,5 +739,3 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_area, size_t src_of
}
return 0;
}
#endif

2
sound/core/oss/pcm_plugin.h
View File

@ -3,7 +3,7 @@
/*
* Digital Audio (Plugin interface) abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

370
sound/core/oss/plugin_ops.h
View File

@ -1,370 +0,0 @@
/*
* Plugin sample operators with fast switch
* Copyright (c) 2000 by Jaroslav Kysela <perex@suse.cz>
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#define as_u8(ptr) (*(u_int8_t*)(ptr))
#define as_u16(ptr) (*(u_int16_t*)(ptr))
#define as_u32(ptr) (*(u_int32_t*)(ptr))
#define as_u64(ptr) (*(u_int64_t*)(ptr))
#define as_s8(ptr) (*(int8_t*)(ptr))
#define as_s16(ptr) (*(int16_t*)(ptr))
#define as_s32(ptr) (*(int32_t*)(ptr))
#define as_s64(ptr) (*(int64_t*)(ptr))
#ifdef COPY_LABELS
static void *copy_labels[4] = {
&&copy_8,
&&copy_16,
&&copy_32,
&&copy_64
};
#endif
#ifdef COPY_END
while(0) {
copy_8: as_s8(dst) = as_s8(src); goto COPY_END;
copy_16: as_s16(dst) = as_s16(src); goto COPY_END;
copy_32: as_s32(dst) = as_s32(src); goto COPY_END;
copy_64: as_s64(dst) = as_s64(src); goto COPY_END;
}
#endif
#ifdef CONV_LABELS
/* src_wid src_endswap sign_toggle dst_wid dst_endswap */
static void *conv_labels[4 * 2 * 2 * 4 * 2] = {
&&conv_xxx1_xxx1, /* 8h -> 8h */
&&conv_xxx1_xxx1, /* 8h -> 8s */
&&conv_xxx1_xx10, /* 8h -> 16h */
&&conv_xxx1_xx01, /* 8h -> 16s */
&&conv_xxx1_x100, /* 8h -> 24h */
&&conv_xxx1_001x, /* 8h -> 24s */
&&conv_xxx1_1000, /* 8h -> 32h */
&&conv_xxx1_0001, /* 8h -> 32s */
&&conv_xxx1_xxx9, /* 8h ^> 8h */
&&conv_xxx1_xxx9, /* 8h ^> 8s */
&&conv_xxx1_xx90, /* 8h ^> 16h */
&&conv_xxx1_xx09, /* 8h ^> 16s */
&&conv_xxx1_x900, /* 8h ^> 24h */
&&conv_xxx1_009x, /* 8h ^> 24s */
&&conv_xxx1_9000, /* 8h ^> 32h */
&&conv_xxx1_0009, /* 8h ^> 32s */
&&conv_xxx1_xxx1, /* 8s -> 8h */
&&conv_xxx1_xxx1, /* 8s -> 8s */
&&conv_xxx1_xx10, /* 8s -> 16h */
&&conv_xxx1_xx01, /* 8s -> 16s */
&&conv_xxx1_x100, /* 8s -> 24h */
&&conv_xxx1_001x, /* 8s -> 24s */
&&conv_xxx1_1000, /* 8s -> 32h */
&&conv_xxx1_0001, /* 8s -> 32s */
&&conv_xxx1_xxx9, /* 8s ^> 8h */
&&conv_xxx1_xxx9, /* 8s ^> 8s */
&&conv_xxx1_xx90, /* 8s ^> 16h */
&&conv_xxx1_xx09, /* 8s ^> 16s */
&&conv_xxx1_x900, /* 8s ^> 24h */
&&conv_xxx1_009x, /* 8s ^> 24s */
&&conv_xxx1_9000, /* 8s ^> 32h */
&&conv_xxx1_0009, /* 8s ^> 32s */
&&conv_xx12_xxx1, /* 16h -> 8h */
&&conv_xx12_xxx1, /* 16h -> 8s */
&&conv_xx12_xx12, /* 16h -> 16h */
&&conv_xx12_xx21, /* 16h -> 16s */
&&conv_xx12_x120, /* 16h -> 24h */
&&conv_xx12_021x, /* 16h -> 24s */
&&conv_xx12_1200, /* 16h -> 32h */
&&conv_xx12_0021, /* 16h -> 32s */
&&conv_xx12_xxx9, /* 16h ^> 8h */
&&conv_xx12_xxx9, /* 16h ^> 8s */
&&conv_xx12_xx92, /* 16h ^> 16h */
&&conv_xx12_xx29, /* 16h ^> 16s */
&&conv_xx12_x920, /* 16h ^> 24h */
&&conv_xx12_029x, /* 16h ^> 24s */
&&conv_xx12_9200, /* 16h ^> 32h */
&&conv_xx12_0029, /* 16h ^> 32s */
&&conv_xx12_xxx2, /* 16s -> 8h */
&&conv_xx12_xxx2, /* 16s -> 8s */
&&conv_xx12_xx21, /* 16s -> 16h */
&&conv_xx12_xx12, /* 16s -> 16s */
&&conv_xx12_x210, /* 16s -> 24h */
&&conv_xx12_012x, /* 16s -> 24s */
&&conv_xx12_2100, /* 16s -> 32h */
&&conv_xx12_0012, /* 16s -> 32s */
&&conv_xx12_xxxA, /* 16s ^> 8h */
&&conv_xx12_xxxA, /* 16s ^> 8s */
&&conv_xx12_xxA1, /* 16s ^> 16h */
&&conv_xx12_xx1A, /* 16s ^> 16s */
&&conv_xx12_xA10, /* 16s ^> 24h */
&&conv_xx12_01Ax, /* 16s ^> 24s */
&&conv_xx12_A100, /* 16s ^> 32h */
&&conv_xx12_001A, /* 16s ^> 32s */
&&conv_x123_xxx1, /* 24h -> 8h */
&&conv_x123_xxx1, /* 24h -> 8s */
&&conv_x123_xx12, /* 24h -> 16h */
&&conv_x123_xx21, /* 24h -> 16s */
&&conv_x123_x123, /* 24h -> 24h */
&&conv_x123_321x, /* 24h -> 24s */
&&conv_x123_1230, /* 24h -> 32h */
&&conv_x123_0321, /* 24h -> 32s */
&&conv_x123_xxx9, /* 24h ^> 8h */
&&conv_x123_xxx9, /* 24h ^> 8s */
&&conv_x123_xx92, /* 24h ^> 16h */
&&conv_x123_xx29, /* 24h ^> 16s */
&&conv_x123_x923, /* 24h ^> 24h */
&&conv_x123_329x, /* 24h ^> 24s */
&&conv_x123_9230, /* 24h ^> 32h */
&&conv_x123_0329, /* 24h ^> 32s */
&&conv_123x_xxx3, /* 24s -> 8h */
&&conv_123x_xxx3, /* 24s -> 8s */
&&conv_123x_xx32, /* 24s -> 16h */
&&conv_123x_xx23, /* 24s -> 16s */
&&conv_123x_x321, /* 24s -> 24h */
&&conv_123x_123x, /* 24s -> 24s */
&&conv_123x_3210, /* 24s -> 32h */
&&conv_123x_0123, /* 24s -> 32s */
&&conv_123x_xxxB, /* 24s ^> 8h */
&&conv_123x_xxxB, /* 24s ^> 8s */
&&conv_123x_xxB2, /* 24s ^> 16h */
&&conv_123x_xx2B, /* 24s ^> 16s */
&&conv_123x_xB21, /* 24s ^> 24h */
&&conv_123x_12Bx, /* 24s ^> 24s */
&&conv_123x_B210, /* 24s ^> 32h */
&&conv_123x_012B, /* 24s ^> 32s */
&&conv_1234_xxx1, /* 32h -> 8h */
&&conv_1234_xxx1, /* 32h -> 8s */
&&conv_1234_xx12, /* 32h -> 16h */
&&conv_1234_xx21, /* 32h -> 16s */
&&conv_1234_x123, /* 32h -> 24h */
&&conv_1234_321x, /* 32h -> 24s */
&&conv_1234_1234, /* 32h -> 32h */
&&conv_1234_4321, /* 32h -> 32s */
&&conv_1234_xxx9, /* 32h ^> 8h */
&&conv_1234_xxx9, /* 32h ^> 8s */
&&conv_1234_xx92, /* 32h ^> 16h */
&&conv_1234_xx29, /* 32h ^> 16s */
&&conv_1234_x923, /* 32h ^> 24h */
&&conv_1234_329x, /* 32h ^> 24s */
&&conv_1234_9234, /* 32h ^> 32h */
&&conv_1234_4329, /* 32h ^> 32s */
&&conv_1234_xxx4, /* 32s -> 8h */
&&conv_1234_xxx4, /* 32s -> 8s */
&&conv_1234_xx43, /* 32s -> 16h */
&&conv_1234_xx34, /* 32s -> 16s */
&&conv_1234_x432, /* 32s -> 24h */
&&conv_1234_234x, /* 32s -> 24s */
&&conv_1234_4321, /* 32s -> 32h */
&&conv_1234_1234, /* 32s -> 32s */
&&conv_1234_xxxC, /* 32s ^> 8h */
&&conv_1234_xxxC, /* 32s ^> 8s */
&&conv_1234_xxC3, /* 32s ^> 16h */
&&conv_1234_xx3C, /* 32s ^> 16s */
&&conv_1234_xC32, /* 32s ^> 24h */
&&conv_1234_23Cx, /* 32s ^> 24s */
&&conv_1234_C321, /* 32s ^> 32h */
&&conv_1234_123C, /* 32s ^> 32s */
};
#endif
#ifdef CONV_END
while(0) {
conv_xxx1_xxx1: as_u8(dst) = as_u8(src); goto CONV_END;
conv_xxx1_xx10: as_u16(dst) = (u_int16_t)as_u8(src) << 8; goto CONV_END;
conv_xxx1_xx01: as_u16(dst) = (u_int16_t)as_u8(src); goto CONV_END;
conv_xxx1_x100: as_u32(dst) = (u_int32_t)as_u8(src) << 16; goto CONV_END;
conv_xxx1_001x: as_u32(dst) = (u_int32_t)as_u8(src) << 8; goto CONV_END;
conv_xxx1_1000: as_u32(dst) = (u_int32_t)as_u8(src) << 24; goto CONV_END;
conv_xxx1_0001: as_u32(dst) = (u_int32_t)as_u8(src); goto CONV_END;
conv_xxx1_xxx9: as_u8(dst) = as_u8(src) ^ 0x80; goto CONV_END;
conv_xxx1_xx90: as_u16(dst) = (u_int16_t)(as_u8(src) ^ 0x80) << 8; goto CONV_END;
conv_xxx1_xx09: as_u16(dst) = (u_int16_t)(as_u8(src) ^ 0x80); goto CONV_END;
conv_xxx1_x900: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 16; goto CONV_END;
conv_xxx1_009x: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 8; goto CONV_END;
conv_xxx1_9000: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 24; goto CONV_END;
conv_xxx1_0009: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80); goto CONV_END;
conv_xx12_xxx1: as_u8(dst) = as_u16(src) >> 8; goto CONV_END;
conv_xx12_xx12: as_u16(dst) = as_u16(src); goto CONV_END;
conv_xx12_xx21: as_u16(dst) = swab16(as_u16(src)); goto CONV_END;
conv_xx12_x120: as_u32(dst) = (u_int32_t)as_u16(src) << 8; goto CONV_END;
conv_xx12_021x: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 8; goto CONV_END;
conv_xx12_1200: as_u32(dst) = (u_int32_t)as_u16(src) << 16; goto CONV_END;
conv_xx12_0021: as_u32(dst) = (u_int32_t)swab16(as_u16(src)); goto CONV_END;
conv_xx12_xxx9: as_u8(dst) = (as_u16(src) >> 8) ^ 0x80; goto CONV_END;
conv_xx12_xx92: as_u16(dst) = as_u16(src) ^ 0x8000; goto CONV_END;
conv_xx12_xx29: as_u16(dst) = swab16(as_u16(src)) ^ 0x80; goto CONV_END;
conv_xx12_x920: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x8000) << 8; goto CONV_END;
conv_xx12_029x: as_u32(dst) = (u_int32_t)(swab16(as_u16(src)) ^ 0x80) << 8; goto CONV_END;
conv_xx12_9200: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x8000) << 16; goto CONV_END;
conv_xx12_0029: as_u32(dst) = (u_int32_t)(swab16(as_u16(src)) ^ 0x80); goto CONV_END;
conv_xx12_xxx2: as_u8(dst) = as_u16(src) & 0xff; goto CONV_END;
conv_xx12_x210: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 8; goto CONV_END;
conv_xx12_012x: as_u32(dst) = (u_int32_t)as_u16(src) << 8; goto CONV_END;
conv_xx12_2100: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 16; goto CONV_END;
conv_xx12_0012: as_u32(dst) = (u_int32_t)as_u16(src); goto CONV_END;
conv_xx12_xxxA: as_u8(dst) = (as_u16(src) ^ 0x80) & 0xff; goto CONV_END;
conv_xx12_xxA1: as_u16(dst) = swab16(as_u16(src) ^ 0x80); goto CONV_END;
conv_xx12_xx1A: as_u16(dst) = as_u16(src) ^ 0x80; goto CONV_END;
conv_xx12_xA10: as_u32(dst) = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 8; goto CONV_END;
conv_xx12_01Ax: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x80) << 8; goto CONV_END;
conv_xx12_A100: as_u32(dst) = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 16; goto CONV_END;
conv_xx12_001A: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x80); goto CONV_END;
conv_x123_xxx1: as_u8(dst) = as_u32(src) >> 16; goto CONV_END;
conv_x123_xx12: as_u16(dst) = as_u32(src) >> 8; goto CONV_END;
conv_x123_xx21: as_u16(dst) = swab16(as_u32(src) >> 8); goto CONV_END;
conv_x123_x123: as_u32(dst) = as_u32(src); goto CONV_END;
conv_x123_321x: as_u32(dst) = swab32(as_u32(src)); goto CONV_END;
conv_x123_1230: as_u32(dst) = as_u32(src) << 8; goto CONV_END;
conv_x123_0321: as_u32(dst) = swab32(as_u32(src)) >> 8; goto CONV_END;
conv_x123_xxx9: as_u8(dst) = (as_u32(src) >> 16) ^ 0x80; goto CONV_END;
conv_x123_xx92: as_u16(dst) = (as_u32(src) >> 8) ^ 0x8000; goto CONV_END;
conv_x123_xx29: as_u16(dst) = swab16(as_u32(src) >> 8) ^ 0x80; goto CONV_END;
conv_x123_x923: as_u32(dst) = as_u32(src) ^ 0x800000; goto CONV_END;
conv_x123_329x: as_u32(dst) = swab32(as_u32(src)) ^ 0x8000; goto CONV_END;
conv_x123_9230: as_u32(dst) = (as_u32(src) ^ 0x800000) << 8; goto CONV_END;
conv_x123_0329: as_u32(dst) = (swab32(as_u32(src)) >> 8) ^ 0x80; goto CONV_END;
conv_123x_xxx3: as_u8(dst) = (as_u32(src) >> 8) & 0xff; goto CONV_END;
conv_123x_xx32: as_u16(dst) = swab16(as_u32(src) >> 8); goto CONV_END;
conv_123x_xx23: as_u16(dst) = (as_u32(src) >> 8) & 0xffff; goto CONV_END;
conv_123x_x321: as_u32(dst) = swab32(as_u32(src)); goto CONV_END;
conv_123x_123x: as_u32(dst) = as_u32(src); goto CONV_END;
conv_123x_3210: as_u32(dst) = swab32(as_u32(src)) << 8; goto CONV_END;
conv_123x_0123: as_u32(dst) = as_u32(src) >> 8; goto CONV_END;
conv_123x_xxxB: as_u8(dst) = ((as_u32(src) >> 8) & 0xff) ^ 0x80; goto CONV_END;
conv_123x_xxB2: as_u16(dst) = swab16((as_u32(src) >> 8) ^ 0x80); goto CONV_END;
conv_123x_xx2B: as_u16(dst) = ((as_u32(src) >> 8) & 0xffff) ^ 0x80; goto CONV_END;
conv_123x_xB21: as_u32(dst) = swab32(as_u32(src)) ^ 0x800000; goto CONV_END;
conv_123x_12Bx: as_u32(dst) = as_u32(src) ^ 0x8000; goto CONV_END;
conv_123x_B210: as_u32(dst) = swab32(as_u32(src) ^ 0x8000) << 8; goto CONV_END;
conv_123x_012B: as_u32(dst) = (as_u32(src) >> 8) ^ 0x80; goto CONV_END;
conv_1234_xxx1: as_u8(dst) = as_u32(src) >> 24; goto CONV_END;
conv_1234_xx12: as_u16(dst) = as_u32(src) >> 16; goto CONV_END;
conv_1234_xx21: as_u16(dst) = swab16(as_u32(src) >> 16); goto CONV_END;
conv_1234_x123: as_u32(dst) = as_u32(src) >> 8; goto CONV_END;
conv_1234_321x: as_u32(dst) = swab32(as_u32(src)) << 8; goto CONV_END;
conv_1234_1234: as_u32(dst) = as_u32(src); goto CONV_END;
conv_1234_4321: as_u32(dst) = swab32(as_u32(src)); goto CONV_END;
conv_1234_xxx9: as_u8(dst) = (as_u32(src) >> 24) ^ 0x80; goto CONV_END;
conv_1234_xx92: as_u16(dst) = (as_u32(src) >> 16) ^ 0x8000; goto CONV_END;
conv_1234_xx29: as_u16(dst) = swab16(as_u32(src) >> 16) ^ 0x80; goto CONV_END;
conv_1234_x923: as_u32(dst) = (as_u32(src) >> 8) ^ 0x800000; goto CONV_END;
conv_1234_329x: as_u32(dst) = (swab32(as_u32(src)) ^ 0x80) << 8; goto CONV_END;
conv_1234_9234: as_u32(dst) = as_u32(src) ^ 0x80000000; goto CONV_END;
conv_1234_4329: as_u32(dst) = swab32(as_u32(src)) ^ 0x80; goto CONV_END;
conv_1234_xxx4: as_u8(dst) = as_u32(src) & 0xff; goto CONV_END;
conv_1234_xx43: as_u16(dst) = swab16(as_u32(src)); goto CONV_END;
conv_1234_xx34: as_u16(dst) = as_u32(src) & 0xffff; goto CONV_END;
conv_1234_x432: as_u32(dst) = swab32(as_u32(src)) >> 8; goto CONV_END;
conv_1234_234x: as_u32(dst) = as_u32(src) << 8; goto CONV_END;
conv_1234_xxxC: as_u8(dst) = (as_u32(src) & 0xff) ^ 0x80; goto CONV_END;
conv_1234_xxC3: as_u16(dst) = swab16(as_u32(src) ^ 0x80); goto CONV_END;
conv_1234_xx3C: as_u16(dst) = (as_u32(src) & 0xffff) ^ 0x80; goto CONV_END;
conv_1234_xC32: as_u32(dst) = (swab32(as_u32(src)) >> 8) ^ 0x800000; goto CONV_END;
conv_1234_23Cx: as_u32(dst) = (as_u32(src) ^ 0x80) << 8; goto CONV_END;
conv_1234_C321: as_u32(dst) = swab32(as_u32(src) ^ 0x80); goto CONV_END;
conv_1234_123C: as_u32(dst) = as_u32(src) ^ 0x80; goto CONV_END;
}
#endif
#ifdef GET_S16_LABELS
/* src_wid src_endswap unsigned */
static void *get_s16_labels[4 * 2 * 2] = {
&&get_s16_xxx1_xx10, /* 8h -> 16h */
&&get_s16_xxx1_xx90, /* 8h ^> 16h */
&&get_s16_xxx1_xx10, /* 8s -> 16h */
&&get_s16_xxx1_xx90, /* 8s ^> 16h */
&&get_s16_xx12_xx12, /* 16h -> 16h */
&&get_s16_xx12_xx92, /* 16h ^> 16h */
&&get_s16_xx12_xx21, /* 16s -> 16h */
&&get_s16_xx12_xxA1, /* 16s ^> 16h */
&&get_s16_x123_xx12, /* 24h -> 16h */
&&get_s16_x123_xx92, /* 24h ^> 16h */
&&get_s16_123x_xx32, /* 24s -> 16h */
&&get_s16_123x_xxB2, /* 24s ^> 16h */
&&get_s16_1234_xx12, /* 32h -> 16h */
&&get_s16_1234_xx92, /* 32h ^> 16h */
&&get_s16_1234_xx43, /* 32s -> 16h */
&&get_s16_1234_xxC3, /* 32s ^> 16h */
};
#endif
#ifdef GET_S16_END
while(0) {
get_s16_xxx1_xx10: sample = (u_int16_t)as_u8(src) << 8; goto GET_S16_END;
get_s16_xxx1_xx90: sample = (u_int16_t)(as_u8(src) ^ 0x80) << 8; goto GET_S16_END;
get_s16_xx12_xx12: sample = as_u16(src); goto GET_S16_END;
get_s16_xx12_xx92: sample = as_u16(src) ^ 0x8000; goto GET_S16_END;
get_s16_xx12_xx21: sample = swab16(as_u16(src)); goto GET_S16_END;
get_s16_xx12_xxA1: sample = swab16(as_u16(src) ^ 0x80); goto GET_S16_END;
get_s16_x123_xx12: sample = as_u32(src) >> 8; goto GET_S16_END;
get_s16_x123_xx92: sample = (as_u32(src) >> 8) ^ 0x8000; goto GET_S16_END;
get_s16_123x_xx32: sample = swab16(as_u32(src) >> 8); goto GET_S16_END;
get_s16_123x_xxB2: sample = swab16((as_u32(src) >> 8) ^ 0x8000); goto GET_S16_END;
get_s16_1234_xx12: sample = as_u32(src) >> 16; goto GET_S16_END;
get_s16_1234_xx92: sample = (as_u32(src) >> 16) ^ 0x8000; goto GET_S16_END;
get_s16_1234_xx43: sample = swab16(as_u32(src)); goto GET_S16_END;
get_s16_1234_xxC3: sample = swab16(as_u32(src) ^ 0x80); goto GET_S16_END;
}
#endif
#ifdef PUT_S16_LABELS
/* dst_wid dst_endswap unsigned */
static void *put_s16_labels[4 * 2 * 2] = {
&&put_s16_xx12_xxx1, /* 16h -> 8h */
&&put_s16_xx12_xxx9, /* 16h ^> 8h */
&&put_s16_xx12_xxx1, /* 16h -> 8s */
&&put_s16_xx12_xxx9, /* 16h ^> 8s */
&&put_s16_xx12_xx12, /* 16h -> 16h */
&&put_s16_xx12_xx92, /* 16h ^> 16h */
&&put_s16_xx12_xx21, /* 16h -> 16s */
&&put_s16_xx12_xx29, /* 16h ^> 16s */
&&put_s16_xx12_x120, /* 16h -> 24h */
&&put_s16_xx12_x920, /* 16h ^> 24h */
&&put_s16_xx12_021x, /* 16h -> 24s */
&&put_s16_xx12_029x, /* 16h ^> 24s */
&&put_s16_xx12_1200, /* 16h -> 32h */
&&put_s16_xx12_9200, /* 16h ^> 32h */
&&put_s16_xx12_0021, /* 16h -> 32s */
&&put_s16_xx12_0029, /* 16h ^> 32s */
};
#endif
#ifdef PUT_S16_END
while (0) {
put_s16_xx12_xxx1: as_u8(dst) = sample >> 8; goto PUT_S16_END;
put_s16_xx12_xxx9: as_u8(dst) = (sample >> 8) ^ 0x80; goto PUT_S16_END;
put_s16_xx12_xx12: as_u16(dst) = sample; goto PUT_S16_END;
put_s16_xx12_xx92: as_u16(dst) = sample ^ 0x8000; goto PUT_S16_END;
put_s16_xx12_xx21: as_u16(dst) = swab16(sample); goto PUT_S16_END;
put_s16_xx12_xx29: as_u16(dst) = swab16(sample) ^ 0x80; goto PUT_S16_END;
put_s16_xx12_x120: as_u32(dst) = (u_int32_t)sample << 8; goto PUT_S16_END;
put_s16_xx12_x920: as_u32(dst) = (u_int32_t)(sample ^ 0x8000) << 8; goto PUT_S16_END;
put_s16_xx12_021x: as_u32(dst) = (u_int32_t)swab16(sample) << 8; goto PUT_S16_END;
put_s16_xx12_029x: as_u32(dst) = (u_int32_t)(swab16(sample) ^ 0x80) << 8; goto PUT_S16_END;
put_s16_xx12_1200: as_u32(dst) = (u_int32_t)sample << 16; goto PUT_S16_END;
put_s16_xx12_9200: as_u32(dst) = (u_int32_t)(sample ^ 0x8000) << 16; goto PUT_S16_END;
put_s16_xx12_0021: as_u32(dst) = (u_int32_t)swab16(sample); goto PUT_S16_END;
put_s16_xx12_0029: as_u32(dst) = (u_int32_t)swab16(sample) ^ 0x80; goto PUT_S16_END;
}
#endif
#undef as_u8
#undef as_u16
#undef as_u32
#undef as_s8
#undef as_s16
#undef as_s32

7
sound/core/oss/rate.c
View File

@ -1,6 +1,6 @@
/*
* Rate conversion Plug-In
* Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
*
*
* This library is free software; you can redistribute it and/or modify
@ -20,9 +20,6 @@
*/
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -340,5 +337,3 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug,
*r_plugin = plugin;
return 0;
}
#endif

5
sound/core/oss/route.c
View File

@ -20,9 +20,6 @@
*/
#include <sound/driver.h>
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
#include <linux/slab.h>
#include <linux/time.h>
#include <sound/core.h>
@ -108,5 +105,3 @@ int snd_pcm_plugin_build_route(struct snd_pcm_substream *plug,
*r_plugin = plugin;
return 0;
}
#endif

4
sound/core/pcm.c
View File

@ -1,6 +1,6 @@
/*
* Digital Audio (PCM) abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@ -30,7 +30,7 @@
#include <sound/control.h>
#include <sound/info.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Abramo Bagnara <abramo@alsa-project.org>");
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>");
MODULE_DESCRIPTION("Midlevel PCM code for ALSA.");
MODULE_LICENSE("GPL");

2
sound/core/pcm_lib.c
View File

@ -1,6 +1,6 @@
/*
* Digital Audio (PCM) abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Abramo Bagnara <abramo@alsa-project.org>
*
*

2
sound/core/pcm_memory.c
View File

@ -1,6 +1,6 @@
/*
* Digital Audio (PCM) abstract layer
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify

65
sound/core/pcm_misc.c
View File

@ -1,6 +1,6 @@
/*
* PCM Interface - misc routines
* Copyright (c) 1998 by Jaroslav Kysela <perex@suse.cz>
* Copyright (c) 1998 by Jaroslav Kysela <perex@perex.cz>
*
*
* This library is free software; you can redistribute it and/or modify
@ -422,38 +422,6 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
EXPORT_SYMBOL(snd_pcm_format_set_silence);
/* [width][unsigned][bigendian] */
static int linear_formats[4][2][2] = {
{{ SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8},
{ SNDRV_PCM_FORMAT_U8, SNDRV_PCM_FORMAT_U8}},
{{SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE},
{SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE}},
{{SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE},
{SNDRV_PCM_FORMAT_U24_LE, SNDRV_PCM_FORMAT_U24_BE}},
{{SNDRV_PCM_FORMAT_S32_LE, SNDRV_PCM_FORMAT_S32_BE},
{SNDRV_PCM_FORMAT_U32_LE, SNDRV_PCM_FORMAT_U32_BE}}
};
/**
* snd_pcm_build_linear_format - return the suitable linear format for the given condition
* @width: the bit-width
* @unsignd: 1 if unsigned, 0 if signed.
* @big_endian: 1 if big-endian, 0 if little-endian
*
* Returns the suitable linear format for the given condition.
*/
snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_endian)
{
if (width & 7)
return SND_PCM_FORMAT_UNKNOWN;
width = (width / 8) - 1;
if (width < 0 || width >= 4)
return SND_PCM_FORMAT_UNKNOWN;
return linear_formats[width][!!unsignd][!!big_endian];
}
EXPORT_SYMBOL(snd_pcm_build_linear_format);
/**
* snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
* @runtime: the runtime instance
@ -465,21 +433,16 @@ EXPORT_SYMBOL(snd_pcm_build_linear_format);
*/
int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
{
static unsigned rates[] = {
/* ATTENTION: these values depend on the definition in pcm.h! */
5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000,
64000, 88200, 96000, 176400, 192000
};
int i;
for (i = 0; i < (int)ARRAY_SIZE(rates); i++) {
for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
if (runtime->hw.rates & (1 << i)) {
runtime->hw.rate_min = rates[i];
runtime->hw.rate_min = snd_pcm_known_rates.list[i];
break;
}
}
for (i = (int)ARRAY_SIZE(rates) - 1; i >= 0; i--) {
for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
if (runtime->hw.rates & (1 << i)) {
runtime->hw.rate_max = rates[i];
runtime->hw.rate_max = snd_pcm_known_rates.list[i];
break;
}
}
@ -487,3 +450,21 @@ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
}
EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
/**
* snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
* @rate: the sample rate to convert
*
* Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or
* SNDRV_PCM_RATE_KNOT for an unknown rate.
*/
unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate)
{
unsigned int i;
for (i = 0; i < snd_pcm_known_rates.count; i++)
if (snd_pcm_known_rates.list[i] == rate)
return 1u << i;
return SNDRV_PCM_RATE_KNOT;
}
EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);

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