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authorLinus Torvalds <torvalds@woody.linux-foundation.org>2007-10-16 10:13:38 -0700
committerLinus Torvalds <torvalds@woody.linux-foundation.org>2007-10-16 10:13:38 -0700
commitfc8a327db6c46de783b1a4276d846841b9abc24c (patch)
treebee512c142cccea93511debd98ef954581693727
parent92d15c2ccbb3e31a3fc71ad28fdb55e1319383c0 (diff)
parent24837e6f249a2c83667552e6871c1543b4a6b934 (diff)
downloadlinux-linaro-stable-fc8a327db6c46de783b1a4276d846841b9abc24c.tar.gz
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 ...
-rw-r--r--CREDITS2
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt115
-rw-r--r--Documentation/sound/alsa/CMIPCI.txt17
-rw-r--r--Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl184
-rw-r--r--Documentation/sound/alsa/OSS-Emulation.txt7
-rw-r--r--Documentation/sound/alsa/hda_codec.txt49
-rw-r--r--Documentation/sound/alsa/powersave.txt41
-rw-r--r--MAINTAINERS6
-rw-r--r--drivers/media/video/cx88/cx88-alsa.c2
-rw-r--r--drivers/net/hp100.c4
-rw-r--r--drivers/pnp/interface.c2
-rw-r--r--drivers/pnp/isapnp/core.c4
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-rw-r--r--include/linux/i2c-id.h1
-rw-r--r--include/linux/spi/at73c213.h25
-rw-r--r--include/sound/ac97_codec.h8
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-rw-r--r--include/sound/ainstr_gf1.h2
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-rw-r--r--include/sound/cs4231-regs.h180
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-rw-r--r--include/sound/driver.h2
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-rw-r--r--include/sound/hdspm.h16
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-rw-r--r--include/sound/mixer_oss.h2
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-rw-r--r--sound/aoa/codecs/snd-aoa-codec-onyx.c20
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-rw-r--r--sound/aoa/fabrics/snd-aoa-fabric-layout.c10
-rw-r--r--sound/arm/sa11xx-uda1341.c35
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-rw-r--r--sound/core/control.c33
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-rw-r--r--sound/i2c/other/tea575x-tuner.c4
-rw-r--r--sound/i2c/tea6330t.c14
-rw-r--r--sound/isa/Kconfig22
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-rw-r--r--sound/isa/cs423x/cs4231.c4
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-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c1
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c22
-rw-r--r--sound/soc/soc-core.c20
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/sparc/cs4231.c805
-rw-r--r--sound/sparc/dbri.c581
-rw-r--r--sound/spi/Kconfig31
-rw-r--r--sound/spi/Makefile5
-rw-r--r--sound/spi/at73c213.c1129
-rw-r--r--sound/spi/at73c213.h119
-rw-r--r--sound/synth/Makefile2
-rw-r--r--sound/synth/emux/Makefile2
-rw-r--r--sound/synth/util_mem.c2
-rw-r--r--sound/usb/Kconfig2
-rw-r--r--sound/usb/caiaq/caiaq-audio.c1
-rw-r--r--sound/usb/caiaq/caiaq-device.c18
-rw-r--r--sound/usb/caiaq/caiaq-device.h1
-rw-r--r--sound/usb/caiaq/caiaq-input.c28
-rw-r--r--sound/usb/usbaudio.c46
-rw-r--r--sound/usb/usbmidi.c46
-rw-r--r--sound/usb/usbmixer.c11
-rw-r--r--sound/usb/usbquirks.h100
370 files changed, 11183 insertions, 5968 deletions
diff --git a/CREDITS b/CREDITS
index b90b4cac6e16..08feda2667d0 100644
--- a/CREDITS
+++ b/CREDITS
@@ -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
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 241e26c4ff92..4b48c2e82c3c 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -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
---------------------
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
index 4b2b15387056..16935c8561f7 100644
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -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
------------------
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 74d3a35b59bc..2c3fc3cb3b6b 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -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);
@@ -3456,6 +3473,13 @@ struct _snd_pcm_runtime {
</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>
callback functions</citetitle></link>.
@@ -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 */
diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt
index bfa0c9aacb4b..022aaeb0e9dd 100644
--- a/Documentation/sound/alsa/OSS-Emulation.txt
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -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.
-
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
index 4eaae2a45534..8e1b02526698 100644
--- a/Documentation/sound/alsa/hda_codec.txt
+++ b/Documentation/sound/alsa/hda_codec.txt
@@ -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.
diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt
new file mode 100644
index 000000000000..9657e8099228
--- /dev/null
+++ b/Documentation/sound/alsa/powersave.txt
@@ -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.
diff --git a/MAINTAINERS b/MAINTAINERS
index c7355e7f09ff..1315cca8fc5f 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -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
diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c
index 90c36c5705c3..141dadf7cf1b 100644
--- a/drivers/media/video/cx88/cx88-alsa.c
+++ b/drivers/media/video/cx88/cx88-alsa.c
@@ -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
diff --git a/drivers/net/hp100.c b/drivers/net/hp100.c
index e4fde17e2841..49421d1cd3a5 100644
--- a/drivers/net/hp100.c
+++ b/drivers/net/hp100.c
@@ -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");
diff --git a/drivers/pnp/interface.c b/drivers/pnp/interface.c
index a0cfb75bbb8d..e0ee28a88da3 100644
--- a/drivers/pnp/interface.c
+++ b/drivers/pnp/interface.c
@@ -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>
*/
diff --git a/drivers/pnp/isapnp/core.c b/drivers/pnp/isapnp/core.c
index b035d60a1dcc..2c925b7cd93e 100644
--- a/drivers/pnp/isapnp/core.c
+++ b/drivers/pnp/isapnp/core.c
@@ -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");
diff --git a/drivers/pnp/isapnp/proc.c b/drivers/pnp/isapnp/proc.c
index 560ccb640816..2b8266c3d40f 100644
--- a/drivers/pnp/isapnp/proc.c
+++ b/drivers/pnp/isapnp/proc.c
@@ -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
diff --git a/drivers/pnp/manager.c b/drivers/pnp/manager.c
index 0826287eef53..ea3eac2404ca 100644
--- a/drivers/pnp/manager.c
+++ b/drivers/pnp/manager.c
@@ -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>
*/
diff --git a/drivers/pnp/resource.c b/drivers/pnp/resource.c
index ef1286900db3..087fed18628f 100644
--- a/drivers/pnp/resource.c
+++ b/drivers/pnp/resource.c
@@ -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>
*/
diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h
index a271b67a8e2d..88c81403eb3f 100644
--- a/include/linux/i2c-id.h
+++ b/include/linux/i2c-id.h
@@ -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 */
diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h
new file mode 100644
index 000000000000..0f20a70e5eb4
--- /dev/null
+++ b/include/linux/spi/at73c213.h
@@ -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 */
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 246ac23534bd..01480581f825 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -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
diff --git a/include/sound/ad1848.h b/include/sound/ad1848.h
index b2c3f00a9b35..d04f9e78c7c1 100644
--- a/include/sound/ad1848.h
+++ b/include/sound/ad1848.h
@@ -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 */
diff --git a/include/sound/ainstr_gf1.h b/include/sound/ainstr_gf1.h
index 47726fe0f46d..b62b665c69c6 100644
--- a/include/sound/ainstr_gf1.h
+++ b/include/sound/ainstr_gf1.h
@@ -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
diff --git a/include/sound/ainstr_iw.h b/include/sound/ainstr_iw.h
index 251feaf1b388..11bd25082600 100644
--- a/include/sound/ainstr_iw.h
+++ b/include/sound/ainstr_iw.h
@@ -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
diff --git a/include/sound/ainstr_simple.h b/include/sound/ainstr_simple.h
index 5eead12e58ae..da08e7287557 100644
--- a/include/sound/ainstr_simple.h
+++ b/include/sound/ainstr_simple.h
@@ -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
diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h
index d647dae912b9..4e80d3fe7381 100644
--- a/include/sound/ak4114.h
+++ b/include/sound/ak4114.h
@@ -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
diff --git a/include/sound/ak4117.h b/include/sound/ak4117.h
index d650d52e3d29..1e8178171baf 100644
--- a/include/sound/ak4117.h
+++ b/include/sound/ak4117.h
@@ -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
diff --git a/include/sound/ak4531_codec.h b/include/sound/ak4531_codec.h
index fb30faab43a8..575296cf7987 100644
--- a/include/sound/ak4531_codec.h
+++ b/include/sound/ak4531_codec.h
@@ -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
diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h
index fd0a6c46f497..891cf1aea8b1 100644
--- a/include/sound/ak4xxx-adda.h
+++ b/include/sound/ak4xxx-adda.h
@@ -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
diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h
index 3f2f4042a20d..64daccbe8b29 100644
--- a/include/sound/asequencer.h
+++ b/include/sound/asequencer.h
@@ -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
diff --git a/include/sound/asound.h b/include/sound/asound.h
index c1621c650a9a..af9d11d315e9 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -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
diff --git a/include/sound/asound_fm.h b/include/sound/asound_fm.h
index 956fdc23c595..8fbcab7cc73b 100644
--- a/include/sound/asound_fm.h
+++ b/include/sound/asound_fm.h
@@ -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
diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h
index 58c9ef3d1825..024ce62f7d16 100644
--- a/include/sound/asoundef.h
+++ b/include/sound/asoundef.h
@@ -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
diff --git a/include/sound/control.h b/include/sound/control.h
index 72e759f619b1..e79baa63912f 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -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 */
diff --git a/include/sound/core.h b/include/sound/core.h
index 4b9e609975ab..6954836487ed 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -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
diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h
new file mode 100644
index 000000000000..f1490265c9b8
--- /dev/null
+++ b/include/sound/cs4231-regs.h
@@ -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 */
diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h
index ab51ce1ba9a5..66055d702aa3 100644
--- a/include/sound/cs4231.h
+++ b/include/sound/cs4231.h
@@ -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)
diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h
index 353910ce9755..6b40ee60f4c5 100644
--- a/include/sound/cs46xx.h
+++ b/include/sound/cs46xx.h
@@ -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
*
diff --git a/include/sound/cs46xx_dsp_scb_types.h b/include/sound/cs46xx_dsp_scb_types.h
index 9cb6c7d09567..080857ad0ca2 100644
--- a/include/sound/cs46xx_dsp_scb_types.h
+++ b/include/sound/cs46xx_dsp_scb_types.h
@@ -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
diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h
index d9da9e59cf37..7c44667e79a6 100644
--- a/include/sound/cs46xx_dsp_spos.h
+++ b/include/sound/cs46xx_dsp_spos.h
@@ -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
diff --git a/include/sound/cs46xx_dsp_task_types.h b/include/sound/cs46xx_dsp_task_types.h
index b3076c487de6..5cf920bfda27 100644
--- a/include/sound/cs46xx_dsp_task_types.h
+++ b/include/sound/cs46xx_dsp_task_types.h
@@ -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
diff --git a/include/sound/cs8403.h b/include/sound/cs8403.h
index c6c3f9f0da78..3a8c174a4209 100644
--- a/include/sound/cs8403.h
+++ b/include/sound/cs8403.h
@@ -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>
*
*
diff --git a/include/sound/cs8427.h b/include/sound/cs8427.h
index 97fd9acf8028..f862cfff5f6a 100644
--- a/include/sound/cs8427.h
+++ b/include/sound/cs8427.h
@@ -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
diff --git a/include/sound/driver.h b/include/sound/driver.h
index 3c522e59a33c..5ccb6c5feecb 100644
--- a/include/sound/driver.h
+++ b/include/sound/driver.h
@@ -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
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 529d0a564367..441aa06dcd6f 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -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 */
diff --git a/include/sound/es1688.h b/include/sound/es1688.h
index fc1c47dae3da..10fcf1465810 100644
--- a/include/sound/es1688.h
+++ b/include/sound/es1688.h
@@ -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
diff --git a/include/sound/gus.h b/include/sound/gus.h
index c49ea57db8cc..e5433d8b78bc 100644
--- a/include/sound/gus.h
+++ b/include/sound/gus.h
@@ -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
diff --git a/include/sound/hda_hwdep.h b/include/sound/hda_hwdep.h
new file mode 100644
index 000000000000..1c0034e87f22
--- /dev/null
+++ b/include/sound/hda_hwdep.h
@@ -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
diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h
index c3c854d99c28..81990b2bcc98 100644
--- a/include/sound/hdspm.h
+++ b/include/sound/hdspm.h
@@ -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] */
diff --git a/include/sound/hwdep.h b/include/sound/hwdep.h
index 94c387b5d724..d9eea013c753 100644
--- a/include/sound/hwdep.h
+++ b/include/sound/hwdep.h
@@ -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
diff --git a/include/sound/info.h b/include/sound/info.h
index 97ffc4fb9969..fecbb1ffd540 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -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
diff --git a/include/sound/initval.h b/include/sound/initval.h
index e85b90750a59..1daa6dff8297 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -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
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 83489c3abbaf..ae2921d9ddcc 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -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
diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h
index 197b9e3d612b..51fbcb4a277a 100644
--- a/include/sound/mixer_oss.h
+++ b/include/sound/mixer_oss.h
@@ -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
diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h
index d5c1396c4c9e..d45218b44dfe 100644
--- a/include/sound/mpu401.h
+++ b/include/sound/mpu401.h
@@ -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
diff --git a/include/sound/opl3.h b/include/sound/opl3.h
index 82fdb0930720..1d14b3f82393 100644
--- a/include/sound/opl3.h
+++ b/include/sound/opl3.h
@@ -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
*
*
diff --git a/include/sound/pcm-indirect.h b/include/sound/pcm-indirect.h
index 7003d7702e26..1df7acaaa535 100644
--- a/include/sound/pcm-indirect.h
+++ b/include/sound/pcm-indirect.h
@@ -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
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 73334e0f823f..5e9cc460075e 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -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)
diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h
index 1cd4f64cdf31..cc4e226f35fd 100644
--- a/include/sound/pcm_oss.h
+++ b/include/sound/pcm_oss.h
@@ -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
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index 7dbcd10fa215..b550a416d075 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -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
diff --git a/include/sound/sb.h b/include/sound/sb.h
index 3ad854b397d2..d0c9ed3546c8 100644
--- a/include/sound/sb.h
+++ b/include/sound/sb.h
@@ -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
diff --git a/include/sound/seq_instr.h b/include/sound/seq_instr.h
index f2db03bfd74e..93b0c51df5b0 100644
--- a/include/sound/seq_instr.h
+++ b/include/sound/seq_instr.h
@@ -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
diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h
index dd789e7cdb20..5efab8b29c57 100644
--- a/include/sound/seq_midi_event.h
+++ b/include/sound/seq_midi_event.h
@@ -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
diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h
index 8d5aea76d7c3..d888433a3096 100644
--- a/include/sound/seq_virmidi.h
+++ b/include/sound/seq_virmidi.h
@@ -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
diff --git a/include/sound/soc.h b/include/sound/soc.h
index db6edba8ef08..f47ef1f75f18 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -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,
diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h
index b5067d3c2387..e8eeb3a1ed29 100644
--- a/include/sound/tea575x-tuner.h
+++ b/include/sound/tea575x-tuner.h
@@ -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
diff --git a/include/sound/timer.h b/include/sound/timer.h
index d42c083db1da..7990469a44ce 100644
--- a/include/sound/timer.h
+++ b/include/sound/timer.h
@@ -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>
*
*
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index d93a96b91875..d136ea2181ed 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -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
diff --git a/include/sound/version.h b/include/sound/version.h
index 6bbcfefd2c38..8d4a8dd89237 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -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)"
diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h
index 203d2b45b788..05ead6698434 100644
--- a/include/sound/ymfpci.h
+++ b/include/sound/ymfpci.h
@@ -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
*
*
diff --git a/sound/Kconfig b/sound/Kconfig
index e48b9b37d228..b2a2db47aff5 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -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"
diff --git a/sound/Makefile b/sound/Makefile
index 3ead922bd9c6..c76d70716fa5 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -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
diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c
index 028852374f21..71e3f9360658 100644
--- a/sound/aoa/codecs/snd-aoa-codec-onyx.c
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
@@ -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
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c
index 2f771f57c76f..70c341684794 100644
--- a/sound/aoa/codecs/snd-aoa-codec-tas.c
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.c
@@ -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)
diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
index 98806283d1b2..8b2ba99d7f8a 100644
--- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c
+++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
@@ -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, \
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
index e7ed868fa7c0..81c64b09d359 100644
--- a/sound/arm/sa11xx-uda1341.c
+++ b/sound/arm/sa11xx-uda1341.c
@@ -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
}
/* }}} */
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 5a01c76d02e8..267039a97bd5 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -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
diff --git a/sound/core/control.c b/sound/core/control.c
index 1f1ab9c1b668..4c3aa8e10378 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -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);
diff --git a/sound/core/device.c b/sound/core/device.c
index 5858b02b0b1d..ea1a0621eefb 100644
--- a/sound/core/device.c
+++ b/sound/core/device.c
@@ -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
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index 51ad95b7c894..bfd9d182b8a3 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -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");
diff --git a/sound/core/info.c b/sound/core/info.c
index bf6dbf99528b..1ffd29bb4cd0 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -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
diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c
index a444bfe2cf74..435c9399f7a9 100644
--- a/sound/core/info_oss.c
+++ b/sound/core/info_oss.c
@@ -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
diff --git a/sound/core/init.c b/sound/core/init.c
index f2fe35737186..2cb7099eb1e1 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -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
diff --git a/sound/core/isadma.c b/sound/core/isadma.c
index d52398727f0a..eb173cef4f05 100644
--- a/sound/core/isadma.c
+++ b/sound/core/isadma.c
@@ -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
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 9b5656d8bcca..9b4992eab479 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -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);
}
diff --git a/sound/core/memory.c b/sound/core/memory.c
index 93537ab7c2ac..25b0f056563e 100644
--- a/sound/core/memory.c
+++ b/sound/core/memory.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* Misc memory accessors
*
diff --git a/sound/core/misc.c b/sound/core/misc.c
index f78cd000e88d..6cabab8cc537 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -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
diff --git a/sound/core/oss/Makefile b/sound/core/oss/Makefile
index e6d5a045ba27..10a79453245f 100644
--- a/sound/core/oss/Makefile
+++ b/sound/core/oss/Makefile
@@ -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
diff --git a/sound/core/oss/copy.c b/sound/core/oss/copy.c
index 6658facc5cda..d6a04c2d5a75 100644
--- a/sound/core/oss/copy.c
+++ b/sound/core/oss/copy.c
@@ -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
diff --git a/sound/core/oss/io.c b/sound/core/oss/io.c
index b6e7ce30e5a3..3ece39fc48db 100644
--- a/sound/core/oss/io.c
+++ b/sound/core/oss/io.c
@@ -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
diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c
index 5b1bcdc64779..06f96a3e86f6 100644
--- a/sound/core/oss/linear.c
+++ b/sound/core/oss/linear.c
@@ -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;
-
- 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
-
- 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;
+ int src_le, dst_le, src_bytes, dst_bytes;
+
+ 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;
+
+ 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
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index fccad8f0a6bb..3ace4a5680ba 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -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);
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 2eb18807e6d0..848db82529ed 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -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
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index fc11572c48cf..d0c4ceb9f0b4 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -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;
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 0e67dd280a5d..14095a927a1b 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -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;
- }
- if (width1 == 32) {
- dwidth1 = -dwidth1;
- width1 = width;
+ 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;
}
- 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
diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h
index 3be91b3d5377..ca2f4c39be46 100644
--- a/sound/core/oss/pcm_plugin.h
+++ b/sound/core/oss/pcm_plugin.h
@@ -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
diff --git a/sound/core/oss/plugin_ops.h b/sound/core/oss/plugin_ops.h
deleted file mode 100644
index 1f5bde4631f1..000000000000
--- a/sound/core/oss/plugin_ops.h
+++ /dev/null
@@ -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
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index 18d8a0f4e816..9eb267913c38 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -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
diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c
index 46917dc0196b..de3ffdeaf7e3 100644
--- a/sound/core/oss/route.c
+++ b/sound/core/oss/route.c
@@ -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
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 2743414fc8fa..cf9b9493d41d 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -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");
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 9fefcaa2c324..806f1fba5446 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -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>
*
*
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 95b1b2f0b1e2..a13e38cfd2c6 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -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
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 0019c59a779d..dd9aa51d8c82 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -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);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 59b29cd482ae..fb3dde4db045 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -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
@@ -1787,12 +1787,18 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100,
48000, 64000, 88200, 96000, 176400, 192000 };
+const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+};
+
static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_pcm_hardware *hw = rule->private;
return snd_interval_list(hw_param_interval(params, rule->var),
- ARRAY_SIZE(rates), rates, hw->rates);
+ snd_pcm_known_rates.count,
+ snd_pcm_known_rates.list, hw->rates);
}
static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params,
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
index d94ed16d21ea..23aa9a27e215 100644
--- a/sound/core/pcm_timer.c
+++ b/sound/core/pcm_timer.c
@@ -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
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index e470c3c7d611..b8e700b94e59 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1,6 +1,6 @@
/*
* 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
@@ -30,14 +30,13 @@
#include <linux/mutex.h>
#include <linux/moduleparam.h>
#include <linux/delay.h>
-#include <linux/wait.h>
#include <sound/rawmidi.h>
#include <sound/info.h>
#include <sound/control.h>
#include <sound/minors.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA.");
MODULE_LICENSE("GPL");
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 402e2b4a34c6..ceef14afee30 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
obj-$(CONFIG_SND) += instr/
diff --git a/sound/core/seq/instr/Makefile b/sound/core/seq/instr/Makefile
index 69138f30a293..608960364813 100644
--- a/sound/core/seq/instr/Makefile
+++ b/sound/core/seq/instr/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
snd-ainstr-fm-objs := ainstr_fm.o
diff --git a/sound/core/seq/instr/ainstr_gf1.c b/sound/core/seq/instr/ainstr_gf1.c
index c640e1cf854d..49400262b1eb 100644
--- a/sound/core/seq/instr/ainstr_gf1.c
+++ b/sound/core/seq/instr/ainstr_gf1.c
@@ -1,6 +1,6 @@
/*
* GF1 (GUS) Patch - Instrument routines
- * 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
* it under the terms of the GNU General Public License as published by
@@ -26,7 +26,7 @@
#include <sound/initval.h>
#include <asm/uaccess.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture GF1 (GUS) Patch support.");
MODULE_LICENSE("GPL");
diff --git a/sound/core/seq/instr/ainstr_iw.c b/sound/core/seq/instr/ainstr_iw.c
index 5367baee2d08..6c40eb73fa9f 100644
--- a/sound/core/seq/instr/ainstr_iw.c
+++ b/sound/core/seq/instr/ainstr_iw.c
@@ -1,6 +1,6 @@
/*
* IWFFFF - AMD InterWave (tm) - Instrument routines
- * 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
* it under the terms of the GNU General Public License as published by
@@ -26,7 +26,7 @@
#include <sound/initval.h>
#include <asm/uaccess.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture IWFFFF support.");
MODULE_LICENSE("GPL");
diff --git a/sound/core/seq/instr/ainstr_simple.c b/sound/core/seq/instr/ainstr_simple.c
index ac717bef9d77..78f68bee24fe 100644
--- a/sound/core/seq/instr/ainstr_simple.c
+++ b/sound/core/seq/instr/ainstr_simple.c
@@ -1,6 +1,6 @@
/*
* Simple (MOD player) - Instrument routines
- * 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
* it under the terms of the GNU General Public License as published by
@@ -26,7 +26,7 @@
#include <sound/initval.h>
#include <asm/uaccess.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture Simple Instrument support.");
MODULE_LICENSE("GPL");
diff --git a/sound/core/seq/oss/Makefile b/sound/core/seq/oss/Makefile
index a37ddedf7107..b38406b8463c 100644
--- a/sound/core/seq/oss/Makefile
+++ b/sound/core/seq/oss/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
snd-seq-oss-objs := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \
diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c
index ca5a2ed4d7c3..d0d721c22eac 100644
--- a/sound/core/seq/oss/seq_oss_init.c
+++ b/sound/core/seq/oss/seq_oss_init.c
@@ -176,29 +176,29 @@ snd_seq_oss_open(struct file *file, int level)
int i, rc;
struct seq_oss_devinfo *dp;
- if ((dp = kzalloc(sizeof(*dp), GFP_KERNEL)) == NULL) {
+ dp = kzalloc(sizeof(*dp), GFP_KERNEL);
+ if (!dp) {
snd_printk(KERN_ERR "can't malloc device info\n");
return -ENOMEM;
}
debug_printk(("oss_open: dp = %p\n", dp));
+ dp->cseq = system_client;
+ dp->port = -1;
+ dp->queue = -1;
+
for (i = 0; i < SNDRV_SEQ_OSS_MAX_CLIENTS; i++) {
if (client_table[i] == NULL)
break;
}
+
+ dp->index = i;
if (i >= SNDRV_SEQ_OSS_MAX_CLIENTS) {
snd_printk(KERN_ERR "too many applications\n");
- kfree(dp);
- return -ENOMEM;
+ rc = -ENOMEM;
+ goto _error;
}
- dp->index = i;
- dp->cseq = system_client;
- dp->port = -1;
- dp->queue = -1;
- dp->readq = NULL;
- dp->writeq = NULL;
-
/* look up synth and midi devices */
snd_seq_oss_synth_setup(dp);
snd_seq_oss_midi_setup(dp);
@@ -211,14 +211,16 @@ snd_seq_oss_open(struct file *file, int level)
/* create port */
debug_printk(("create new port\n"));
- if ((rc = create_port(dp)) < 0) {
+ rc = create_port(dp);
+ if (rc < 0) {
snd_printk(KERN_ERR "can't create port\n");
goto _error;
}
/* allocate queue */
debug_printk(("allocate queue\n"));
- if ((rc = alloc_seq_queue(dp)) < 0)
+ rc = alloc_seq_queue(dp);
+ if (rc < 0)
goto _error;
/* set address */
@@ -235,7 +237,8 @@ snd_seq_oss_open(struct file *file, int level)
/* initialize read queue */
debug_printk(("initialize read queue\n"));
if (is_read_mode(dp->file_mode)) {
- if ((dp->readq = snd_seq_oss_readq_new(dp, maxqlen)) == NULL) {
+ dp->readq = snd_seq_oss_readq_new(dp, maxqlen);
+ if (!dp->readq) {
rc = -ENOMEM;
goto _error;
}
@@ -245,7 +248,7 @@ snd_seq_oss_open(struct file *file, int level)
debug_printk(("initialize write queue\n"));
if (is_write_mode(dp->file_mode)) {
dp->writeq = snd_seq_oss_writeq_new(dp, maxqlen);
- if (dp->writeq == NULL) {
+ if (!dp->writeq) {
rc = -ENOMEM;
goto _error;
}
@@ -253,7 +256,8 @@ snd_seq_oss_open(struct file *file, int level)
/* initialize timer */
debug_printk(("initialize timer\n"));
- if ((dp->timer = snd_seq_oss_timer_new(dp)) == NULL) {
+ dp->timer = snd_seq_oss_timer_new(dp);
+ if (!dp->timer) {
snd_printk(KERN_ERR "can't alloc timer\n");
rc = -ENOMEM;
goto _error;
@@ -276,11 +280,13 @@ snd_seq_oss_open(struct file *file, int level)
return 0;
_error:
+ snd_seq_oss_writeq_delete(dp->writeq);
+ snd_seq_oss_readq_delete(dp->readq);
snd_seq_oss_synth_cleanup(dp);
snd_seq_oss_midi_cleanup(dp);
- i = dp->queue;
delete_port(dp);
- delete_seq_queue(i);
+ delete_seq_queue(dp->queue);
+ kfree(dp);
return rc;
}
diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c
index 5c8495601a38..217424858191 100644
--- a/sound/core/seq/oss/seq_oss_writeq.c
+++ b/sound/core/seq/oss/seq_oss_writeq.c
@@ -63,8 +63,10 @@ snd_seq_oss_writeq_new(struct seq_oss_devinfo *dp, int maxlen)
void
snd_seq_oss_writeq_delete(struct seq_oss_writeq *q)
{
- snd_seq_oss_writeq_clear(q); /* to be sure */
- kfree(q);
+ if (q) {
+ snd_seq_oss_writeq_clear(q); /* to be sure */
+ kfree(q);
+ }
}
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index 2f0d8773ac6b..1878208a8026 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -53,7 +53,7 @@ int seq_default_timer_device =
int seq_default_timer_subdevice = 0;
int seq_default_timer_resolution = 0; /* Hz */
-MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer.");
MODULE_LICENSE("GPL");
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index b31b5282a2c8..2e3fa25ab19f 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -1,7 +1,7 @@
/*
* ALSA sequencer Client Manager
* Copyright (c) 1998-2001 by Frank van de Pol <fvdpol@coil.demon.nl>
- * Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
* Takashi Iwai <tiwai@suse.de>
*
*
diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c
index 5efe6523a589..9a6fd56c9109 100644
--- a/sound/core/seq/seq_instr.c
+++ b/sound/core/seq/seq_instr.c
@@ -1,6 +1,6 @@
/*
* Generic Instrument routines for 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
* it under the terms of the GNU General Public License as published by
@@ -26,7 +26,7 @@
#include <sound/seq_instr.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer instrument library.");
MODULE_LICENSE("GPL");
@@ -109,7 +109,7 @@ void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr)
spin_lock_irqsave(&list->lock, flags);
while (instr->use) {
spin_unlock_irqrestore(&list->lock, flags);
- schedule_timeout(1);
+ schedule_timeout_uninterruptible(1);
spin_lock_irqsave(&list->lock, flags);
}
spin_unlock_irqrestore(&list->lock, flags);
@@ -198,8 +198,10 @@ int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list,
while (flist) {
instr = flist;
flist = instr->next;
- while (instr->use)
- schedule_timeout(1);
+ while (instr->use) {
+ schedule_timeout_uninterruptible(1);
+ barrier();
+ }
if (snd_seq_instr_free(instr, atomic)<0)
snd_printk(KERN_WARNING "instrument free problem\n");
instr = next;
@@ -555,7 +557,7 @@ static int instr_free(struct snd_seq_kinstr_ops *ops,
SNDRV_SEQ_INSTR_NOTIFY_REMOVE);
while (instr->use) {
spin_unlock_irqrestore(&list->lock, flags);
- schedule_timeout(1);
+ schedule_timeout_uninterruptible(1);
spin_lock_irqsave(&list->lock, flags);
}
spin_unlock_irqrestore(&list->lock, flags);
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index a3dc5e01e9f2..a72a1945bf8a 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -1,7 +1,7 @@
/*
* ALSA sequencer Memory Manager
* Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
- * Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
* 2000 by Takashi Iwai <tiwai@suse.de>
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c
index 1daa5b069c79..5929aaf1df9d 100644
--- a/sound/core/seq/seq_midi.c
+++ b/sound/core/seq/seq_midi.c
@@ -1,7 +1,7 @@
/*
* Generic MIDI synth driver for ALSA sequencer
* Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
- * 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
@@ -40,7 +40,7 @@ Possible options for midisynth module:
#include <sound/seq_midi_event.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI synth.");
MODULE_LICENSE("GPL");
static int output_buffer_size = PAGE_SIZE;
diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c
index 5ff80b776906..b6820a5a73fc 100644
--- a/sound/core/seq/seq_midi_event.c
+++ b/sound/core/seq/seq_midi_event.c
@@ -2,7 +2,7 @@
* 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
@@ -28,14 +28,13 @@
#include <sound/seq_midi_event.h>
#include <sound/asoundef.h>
-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("MIDI byte <-> sequencer event coder");
MODULE_LICENSE("GPL");
-/* queue type */
-/* from 0 to 7 are normal commands (note off, on, etc.) */
-#define ST_NOTEOFF 0
-#define ST_NOTEON 1
+/* event type, index into status_event[] */
+/* from 0 to 6 are normal commands (note off, on, etc.) for 0x9?-0xe? */
+#define ST_INVALID 7
#define ST_SPECIAL 8
#define ST_SYSEX ST_SPECIAL
/* from 8 to 15 are events for 0xf0-0xf7 */
@@ -65,32 +64,33 @@ static struct status_event_list {
void (*encode)(struct snd_midi_event *dev, struct snd_seq_event *ev);
void (*decode)(struct snd_seq_event *ev, unsigned char *buf);
} status_event[] = {
- /* 0x80 - 0xf0 */
- {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode},
- {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode},
- {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode},
- {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode},
- {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode},
- {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode},
- {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode},
- {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf0 */
+ /* 0x80 - 0xef */
+ {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode},
+ {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode},
+ {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode},
+ {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode},
+ {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode},
+ {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode},
+ {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode},
+ /* invalid */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL},
/* 0xf0 - 0xff */
- {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */
- {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */
- {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */
- {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */
- {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf4 */
- {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf5 */
- {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */
- {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf7 */
- {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */
- {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf9 */
- {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */
- {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */
- {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */
- {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xfd */
- {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */
- {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */
+ {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */
+ {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */
+ {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */
+ {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf4 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf5 */
+ {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf7 */
+ {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf9 */
+ {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */
+ {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */
+ {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xfd */
+ {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */
+ {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */
};
static int extra_decode_ctrl14(struct snd_midi_event *dev, unsigned char *buf, int len,
@@ -129,6 +129,7 @@ int snd_midi_event_new(int bufsize, struct snd_midi_event **rdev)
}
dev->bufsize = bufsize;
dev->lastcmd = 0xff;
+ dev->type = ST_INVALID;
spin_lock_init(&dev->lock);
*rdev = dev;
return 0;
@@ -149,7 +150,7 @@ static inline void reset_encode(struct snd_midi_event *dev)
{
dev->read = 0;
dev->qlen = 0;
- dev->type = 0;
+ dev->type = ST_INVALID;
}
void snd_midi_event_reset_encode(struct snd_midi_event *dev)
@@ -251,29 +252,31 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c,
ev->type = status_event[ST_SPECIAL + c - 0xf0].event;
ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK;
ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED;
- return 1;
+ return ev->type != SNDRV_SEQ_EVENT_NONE;
}
spin_lock_irqsave(&dev->lock, flags);
- if (dev->qlen > 0) {
- /* rest of command */
- dev->buf[dev->read++] = c;
- if (dev->type != ST_SYSEX)
- dev->qlen--;
- } else {
+ if ((c & 0x80) &&
+ (c != MIDI_CMD_COMMON_SYSEX_END || dev->type != ST_SYSEX)) {
/* new command */
+ dev->buf[0] = c;
+ if ((c & 0xf0) == 0xf0) /* system messages */
+ dev->type = (c & 0x0f) + ST_SPECIAL;
+ else
+ dev->type = (c >> 4) & 0x07;
dev->read = 1;
- if (c & 0x80) {
- dev->buf[0] = c;
- if ((c & 0xf0) == 0xf0) /* special events */
- dev->type = (c & 0x0f) + ST_SPECIAL;
- else
- dev->type = (c >> 4) & 0x07;
- dev->qlen = status_event[dev->type].qlen;
- } else {
- /* process this byte as argument */
+ dev->qlen = status_event[dev->type].qlen;
+ } else {
+ if (dev->qlen > 0) {
+ /* rest of command */
dev->buf[dev->read++] = c;
+ if (dev->type != ST_SYSEX)
+ dev->qlen--;
+ } else {
+ /* running status */
+ dev->buf[1] = c;
dev->qlen = status_event[dev->type].qlen - 1;
+ dev->read = 2;
}
}
if (dev->qlen == 0) {
@@ -282,6 +285,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c,
ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED;
if (status_event[dev->type].encode) /* set data values */
status_event[dev->type].encode(dev, ev);
+ if (dev->type >= ST_SPECIAL)
+ dev->type = ST_INVALID;
rc = 1;
} else if (dev->type == ST_SYSEX) {
if (c == MIDI_CMD_COMMON_SYSEX_END ||
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index eefd1cf872b4..b6e23ad12ab9 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -1,7 +1,7 @@
/*
* ALSA sequencer Ports
* Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
- * Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index b4b9a132cb16..8716352afc81 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -1,7 +1,7 @@
/*
* ALSA sequencer Timer
* Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
- * Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 8dc7a3b32b98..7b486c4d70db 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -1,6 +1,6 @@
/*
* Advanced Linux Sound Architecture
- * 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
@@ -42,7 +42,7 @@ EXPORT_SYMBOL(snd_major);
static int cards_limit = 1;
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture driver for soundcards.");
MODULE_LICENSE("GPL");
module_param(major, int, 0444);
@@ -266,6 +266,14 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
snd_minors[minor] = preg;
preg->dev = device_create(sound_class, device, MKDEV(major, minor),
"%s", name);
+ if (IS_ERR(preg->dev)) {
+ snd_minors[minor] = NULL;
+ mutex_unlock(&sound_mutex);
+ minor = PTR_ERR(preg->dev);
+ kfree(preg);
+ return minor;
+ }
+
if (preg->dev)
dev_set_drvdata(preg->dev, private_data);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 4566df41912a..dc73313b733a 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -1,6 +1,6 @@
/*
* Advanced Linux Sound Architecture
- * 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
diff --git a/sound/core/timer.c b/sound/core/timer.c
index f2bbacedd567..e7dc56ca4b97 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1,6 +1,6 @@
/*
* Timers 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
@@ -44,7 +44,7 @@
#endif
static int timer_limit = DEFAULT_TIMER_LIMIT;
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>");
MODULE_DESCRIPTION("ALSA timer interface");
MODULE_LICENSE("GPL");
module_param(timer_limit, int, 0444);
diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile
index 04112642611a..80aeff5ccdea 100644
--- a/sound/drivers/Makefile
+++ b/sound/drivers/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-dummy-objs := dummy.o
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 4360ae9de19c..e008f3c58eac 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1,6 +1,6 @@
/*
* Dummy soundcard
- * 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
@@ -34,7 +34,7 @@
#include <sound/rawmidi.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Dummy soundcard (/dev/null)");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}");
@@ -510,15 +510,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_dummy, -4500, 30, 0);
.get = snd_dummy_capsrc_get, .put = snd_dummy_capsrc_put, \
.private_value = addr }
-static int snd_dummy_capsrc_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 snd_dummy_capsrc_info snd_ctl_boolean_stereo_info
static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/drivers/mpu401/Makefile b/sound/drivers/mpu401/Makefile
index 3fe185d19ae5..918f83f34c11 100644
--- a/sound/drivers/mpu401/Makefile
+++ b/sound/drivers/mpu401/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-mpu401-objs := mpu401.o
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 67c6e9745418..1fc95dadde1d 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -1,6 +1,6 @@
/*
* Driver for generic MPU-401 boards (UART mode only)
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Copyright (c) 2004 by Castet Matthieu <castet.matthieu@free.fr>
*
*
@@ -30,7 +30,7 @@
#include <sound/mpu401.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("MPU-401 UART");
MODULE_LICENSE("GPL");
@@ -70,6 +70,9 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard)
struct snd_card *card;
int err;
+ if (!uart_enter[dev])
+ snd_printk(KERN_ERR "the uart_enter option is obsolete; remove it\n");
+
*rcard = NULL;
card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
if (card == NULL)
@@ -83,8 +86,7 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard)
strcat(card->longname, "polled");
}
- err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev],
- uart_enter[dev] ? 0 : MPU401_INFO_UART_ONLY,
+ err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0,
irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0,
NULL);
if (err < 0) {
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 85aedc348e2d..3306ecd49243 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of MPU-401 in UART mode
*
* MPU-401 supports UART mode which is not capable generate transmit
@@ -39,7 +39,7 @@
#include <sound/core.h>
#include <sound/mpu401.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of MPU-401 in UART mode");
MODULE_LICENSE("GPL");
@@ -270,8 +270,7 @@ static int snd_mpu401_do_reset(struct snd_mpu401 *mpu)
{
if (snd_mpu401_uart_cmd(mpu, MPU401_RESET, 1))
return -EIO;
- if (!(mpu->info_flags & MPU401_INFO_UART_ONLY) &&
- snd_mpu401_uart_cmd(mpu, MPU401_ENTER_UART, 1))
+ if (snd_mpu401_uart_cmd(mpu, MPU401_ENTER_UART, 0))
return -EIO;
return 0;
}
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
index 2025db5947ae..911c159bb3d3 100644
--- a/sound/drivers/mts64.c
+++ b/sound/drivers/mts64.c
@@ -440,15 +440,7 @@ static void mts64_write_midi(struct mts64 *mts, u8 c,
*********************************************************************/
/* SMPTE Switch */
-static int snd_mts64_ctl_smpte_switch_info(struct snd_kcontrol *kctl,
- 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 snd_mts64_ctl_smpte_switch_info snd_ctl_boolean_mono_info
static int snd_mts64_ctl_smpte_switch_get(struct snd_kcontrol* kctl,
struct snd_ctl_elem_value *uctl)
diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile
index 12059785b5cb..19767a6a5c54 100644
--- a/sound/drivers/opl3/Makefile
+++ b/sound/drivers/opl3/Makefile
@@ -1,13 +1,11 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-opl3-lib-objs := opl3_lib.o opl3_synth.o
-snd-opl3-synth-objs := opl3_seq.o opl3_midi.o opl3_drums.o
-ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-snd-opl3-synth-objs += opl3_oss.o
-endif
+snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o
+snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o
#
# this function returns:
diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c
index 87fe376f38f0..a2b9ce060295 100644
--- a/sound/drivers/opl3/opl3_lib.c
+++ b/sound/drivers/opl3/opl3_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Hannu Savolainen 1993-1996,
* Rob Hooft
*
@@ -31,7 +31,7 @@
#include <linux/ioport.h>
#include <sound/minors.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Hannu Savolainen 1993-1996, Rob Hooft");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Hannu Savolainen 1993-1996, Rob Hooft");
MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)");
MODULE_LICENSE("GPL");
diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile
index 141aacbaf315..d178b39ffa60 100644
--- a/sound/drivers/opl4/Makefile
+++ b/sound/drivers/opl4/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index d3e6a20edd38..65de3a755ddb 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -1,6 +1,6 @@
/*
* serial.c
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Isaku Yamahata <yamahata@private.email.ne.jp>,
* George Hansper <ghansper@apana.org.au>,
* Hannu Savolainen
diff --git a/sound/drivers/vx/Makefile b/sound/drivers/vx/Makefile
index 269bd8544a5d..9a168a3c1560 100644
--- a/sound/drivers/vx/Makefile
+++ b/sound/drivers/vx/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-vx-lib-objs := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o
diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c
index f63152a6a223..b8fcd79a7e11 100644
--- a/sound/drivers/vx/vx_mixer.c
+++ b/sound/drivers/vx/vx_mixer.c
@@ -647,14 +647,7 @@ static int vx_audio_monitor_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
return 0;
}
-static int vx_audio_sw_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 vx_audio_sw_info snd_ctl_boolean_stereo_info
static int vx_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -865,14 +858,7 @@ static int vx_peak_meter_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
return 0;
}
-static int vx_saturation_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 vx_saturation_info snd_ctl_boolean_stereo_info
static int vx_saturation_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile
index 45902d48c89c..37970666a453 100644
--- a/sound/i2c/Makefile
+++ b/sound/i2c/Makefile
@@ -1,15 +1,13 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-i2c-objs := i2c.o
snd-cs8427-objs := cs8427.o
snd-tea6330t-objs := tea6330t.o
-ifeq ($(subst m,y,$(CONFIG_L3)),y)
- obj-$(CONFIG_L3) += l3/
-endif
+obj-$(CONFIG_L3) += l3/
obj-$(CONFIG_SND) += other/
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index 64388cb8d6e5..744366b72345 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -1,7 +1,7 @@
/*
* Routines for control of the CS8427 via i2c bus
* IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic
- * 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
@@ -32,7 +32,7 @@
static void snd_cs8427_reset(struct snd_i2c_device *cs8427);
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic");
MODULE_LICENSE("GPL");
@@ -229,6 +229,12 @@ int snd_cs8427_create(struct snd_i2c_bus *bus,
snd_i2c_lock(bus);
err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER);
if (err != CS8427_VER8427A) {
+ /* give second chance */
+ snd_printk(KERN_WARNING "invalid CS8427 signature 0x%x: "
+ "let me try again...\n", err);
+ err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER);
+ }
+ if (err != CS8427_VER8427A) {
snd_i2c_unlock(bus);
snd_printk(KERN_ERR "unable to find CS8427 signature "
"(expected 0x%x, read 0x%x),\n",
diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c
index b60fb1892828..1e58a963b2a7 100644
--- a/sound/i2c/i2c.c
+++ b/sound/i2c/i2c.c
@@ -2,7 +2,7 @@
* Generic i2c interface for ALSA
*
* (c) 1998 Gerd Knorr <kraxel@cs.tu-berlin.de>
- * Modified for the ALSA driver by Jaroslav Kysela <perex@suse.cz>
+ * Modified for the ALSA driver 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
@@ -28,7 +28,7 @@
#include <sound/core.h>
#include <sound/i2c.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Generic i2c interface for ALSA");
MODULE_LICENSE("GPL");
diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile
index 77a8a7c75dd9..703d954238f4 100644
--- a/sound/i2c/other/Makefile
+++ b/sound/i2c/other/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2003 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
#
snd-ak4114-objs := ak4114.o
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
index 1efb973137a6..facde46f957a 100644
--- a/sound/i2c/other/ak4114.c
+++ b/sound/i2c/other/ak4114.c
@@ -1,7 +1,7 @@
/*
* Routines for control of the AK4114 via I2C and 4-wire serial interface
* IEC958 (S/PDIF) receiver by Asahi Kasei
- * 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
@@ -29,7 +29,7 @@
#include <sound/ak4114.h>
#include <sound/asoundef.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei");
MODULE_LICENSE("GPL");
@@ -200,15 +200,7 @@ static int snd_ak4114_in_error_get(struct snd_kcontrol *kcontrol,
return 0;
}
-static int snd_ak4114_in_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 snd_ak4114_in_bit_info snd_ctl_boolean_mono_info
static int snd_ak4114_in_bit_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c
index c022f29da2f7..ee1585aec99b 100644
--- a/sound/i2c/other/ak4117.c
+++ b/sound/i2c/other/ak4117.c
@@ -1,7 +1,7 @@
/*
* Routines for control of the AK4117 via 4-wire serial interface
* IEC958 (S/PDIF) receiver by Asahi Kasei
- * 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
@@ -29,7 +29,7 @@
#include <sound/ak4117.h>
#include <sound/asoundef.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("AK4117 IEC958 (S/PDIF) receiver by Asahi Kasei");
MODULE_LICENSE("GPL");
@@ -181,15 +181,7 @@ static int snd_ak4117_in_error_get(struct snd_kcontrol *kcontrol,
return 0;
}
-static int snd_ak4117_in_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 snd_ak4117_in_bit_info snd_ctl_boolean_mono_info
static int snd_ak4117_in_bit_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index fd335159f849..de03f689fa2e 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -2,7 +2,7 @@
* ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4358 / AK4381
* AD and DA converters
*
- * Copyright (c) 2000-2004 Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) 2000-2004 Jaroslav Kysela <perex@perex.cz>,
* Takashi Iwai <tiwai@suse.de>
*
* This program is free software; you can redistribute it and/or modify
@@ -31,7 +31,7 @@
#include <sound/tlv.h>
#include <sound/ak4xxx-adda.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>");
MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters");
MODULE_LICENSE("GPL");
@@ -463,15 +463,7 @@ static int snd_akm4xxx_deemphasis_put(struct snd_kcontrol *kcontrol,
return change;
}
-static int ak4xxx_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 ak4xxx_switch_info snd_ctl_boolean_mono_info
static int ak4xxx_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c
index e91cc3b44de5..00c83d8b32b1 100644
--- a/sound/i2c/other/pt2258.c
+++ b/sound/i2c/other/pt2258.c
@@ -140,15 +140,7 @@ static int pt2258_stereo_volume_put(struct snd_kcontrol *kcontrol,
return -EIO;
}
-static int pt2258_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 pt2258_switch_info snd_ctl_boolean_mono_info
static int pt2258_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index 4c2fd14c1056..fe31bb5cffb8 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -1,7 +1,7 @@
/*
* ALSA driver for TEA5757/5759 Philips AM/FM radio 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
@@ -28,7 +28,7 @@
#include <sound/core.h>
#include <sound/tea575x-tuner.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
MODULE_LICENSE("GPL");
diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c
index ae5b1e3a68ce..9bab744af0ef 100644
--- a/sound/i2c/tea6330t.c
+++ b/sound/i2c/tea6330t.c
@@ -1,7 +1,7 @@
/*
* Routines for control of the TEA6330T circuit via i2c bus
* Sound fader control circuit for car radios by Philips Semiconductors
- * 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
@@ -27,7 +27,7 @@
#include <sound/control.h>
#include <sound/tea6330t.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of the TEA6330T circuit via i2c bus");
MODULE_LICENSE("GPL");
@@ -142,15 +142,7 @@ static int snd_tea6330t_put_master_volume(struct snd_kcontrol *kcontrol,
.info = snd_tea6330t_info_master_switch, \
.get = snd_tea6330t_get_master_switch, .put = snd_tea6330t_put_master_switch }
-static int snd_tea6330t_info_master_switch(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 snd_tea6330t_info_master_switch snd_ctl_boolean_stereo_info
static int snd_tea6330t_get_master_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index ea5084abe60f..2639a6ab8f2e 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -191,6 +191,19 @@ config SND_ES18XX
To compile this driver as a module, choose M here: the module
will be called snd-es18xx.
+config SND_SC6000
+ tristate "Gallant SC-6000, Audio Excel DSP 16"
+ depends on SND && HAS_IOPORT
+ select SND_AD1848_LIB
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ help
+ Say Y here to include support for Gallant SC-6000 card and clones:
+ Audio Excel DSP 16 and Zoltrix AV302.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sc6000.
+
config SND_GUS_SYNTH
tristate
@@ -414,7 +427,7 @@ config SND_SSCAPE
config SND_WAVEFRONT
tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)"
depends on SND
- select FW_LOADER if !SND_WAVEFRONT_FIRMWARE_IN_KERNEL
+ select FW_LOADER
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_CS4231_LIB
@@ -430,8 +443,9 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL
depends on SND_WAVEFRONT
default y
help
- Say Y here to include the static firmware built in the kernel
- for the Wavefront driver. If you choose N here, you need to
- install the firmware files from the alsa-firmware package.
+ Say Y here to include the static firmware for FX DSP built in
+ the kernel for the Wavefront driver. If you choose N here,
+ you need to install the firmware files from the
+ alsa-firmware package.
endmenu
diff --git a/sound/isa/Makefile b/sound/isa/Makefile
index bb317ccc170f..c0ce7db2a1b5 100644
--- a/sound/isa/Makefile
+++ b/sound/isa/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-adlib-objs := adlib.o
@@ -10,6 +10,7 @@ snd-cmi8330-objs := cmi8330.o
snd-dt019x-objs := dt019x.o
snd-es18xx-objs := es18xx.o
snd-opl3sa2-objs := opl3sa2.o
+snd-sc6000-objs := sc6000.o
snd-sgalaxy-objs := sgalaxy.o
snd-sscape-objs := sscape.o
@@ -21,6 +22,7 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o
obj-$(CONFIG_SND_DT019X) += snd-dt019x.o
obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o
obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o
+obj-$(CONFIG_SND_SC6000) += snd-sc6000.o
obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o
obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o
diff --git a/sound/isa/ad1816a/Makefile b/sound/isa/ad1816a/Makefile
index 90e00e842e49..487ab23860e3 100644
--- a/sound/isa/ad1816a/Makefile
+++ b/sound/isa/ad1816a/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ad1816a-objs := ad1816a.o ad1816a_lib.o
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
index ec9209cd5177..cf18fe4617a1 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -453,7 +453,6 @@ static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream)
if ((error = snd_ad1816a_open(chip, AD1816A_MODE_PLAYBACK)) < 0)
return error;
- snd_pcm_set_sync(substream);
runtime->hw = snd_ad1816a_playback;
snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max);
snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max);
@@ -469,7 +468,6 @@ static int snd_ad1816a_capture_open(struct snd_pcm_substream *substream)
if ((error = snd_ad1816a_open(chip, AD1816A_MODE_CAPTURE)) < 0)
return error;
- snd_pcm_set_sync(substream);
runtime->hw = snd_ad1816a_capture;
snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max);
snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max);
diff --git a/sound/isa/ad1848/Makefile b/sound/isa/ad1848/Makefile
index 45d59998aa69..ae23331e9200 100644
--- a/sound/isa/ad1848/Makefile
+++ b/sound/isa/ad1848/Makefile
@@ -1,15 +1,12 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ad1848-lib-objs := ad1848_lib.o
snd-ad1848-objs := ad1848.o
# Toplevel Module Dependency
-obj-$(CONFIG_SND_CMI8330) += snd-ad1848-lib.o
-obj-$(CONFIG_SND_SGALAXY) += snd-ad1848-lib.o
-obj-$(CONFIG_SND_AD1848) += snd-ad1848.o snd-ad1848-lib.o
-obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-ad1848-lib.o
+obj-$(CONFIG_SND_AD1848) += snd-ad1848.o
+obj-$(CONFIG_SND_AD1848_LIB) += snd-ad1848-lib.o
-obj-m := $(sort $(obj-m))
diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c
index d09a7fa86545..a4710b5e214c 100644
--- a/sound/isa/ad1848/ad1848.c
+++ b/sound/isa/ad1848/ad1848.c
@@ -1,8 +1,8 @@
/*
* Generic driver for AD1848/AD1847/CS4248 chips (0.1 Alpha)
* Copyright (c) by Tugrul Galatali <galatalt@stuy.edu>,
- * Jaroslav Kysela <perex@suse.cz>
- * Based on card-4232.c by Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
+ * Based on card-4232.c by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@@ -36,7 +36,7 @@
#define DEV_NAME "ad1848"
MODULE_DESCRIPTION(CRD_NAME);
-MODULE_AUTHOR("Tugrul Galatali <galatalt@stuy.edu>, Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Tugrul Galatali <galatalt@stuy.edu>, Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1848},"
"{Analog Devices,AD1847},"
diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c
index 1bc2e3fd5721..a901cd1ee692 100644
--- a/sound/isa/ad1848/ad1848_lib.c
+++ b/sound/isa/ad1848/ad1848_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of AD1848/AD1847/CS4248
*
*
@@ -35,7 +35,7 @@
#include <asm/io.h>
#include <asm/dma.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of AD1848/AD1847/CS4248");
MODULE_LICENSE("GPL");
@@ -70,7 +70,7 @@ static unsigned int rates[14] = {
};
static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
- .count = 14,
+ .count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
@@ -99,24 +99,32 @@ static unsigned char snd_ad1848_original_image[16] =
* Basic I/O functions
*/
-void snd_ad1848_out(struct snd_ad1848 *chip,
- unsigned char reg,
- unsigned char value)
+static void snd_ad1848_wait(struct snd_ad1848 *chip)
{
int timeout;
- for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--)
+ for (timeout = 250; timeout > 0; timeout--) {
+ if ((inb(AD1848P(chip, REGSEL)) & AD1848_INIT) == 0)
+ break;
udelay(100);
+ }
+}
+
+void snd_ad1848_out(struct snd_ad1848 *chip,
+ unsigned char reg,
+ unsigned char value)
+{
+ snd_ad1848_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT)
- snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
+ snd_printk(KERN_WARNING "auto calibration time out - "
+ "reg = 0x%x, value = 0x%x\n", reg, value);
#endif
outb(chip->mce_bit | reg, AD1848P(chip, REGSEL));
outb(chip->image[reg] = value, AD1848P(chip, REG));
mb();
-#if 0
- printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value);
-#endif
+ snd_printdd("codec out - reg 0x%x = 0x%x\n",
+ chip->mce_bit | reg, value);
}
EXPORT_SYMBOL(snd_ad1848_out);
@@ -124,10 +132,7 @@ EXPORT_SYMBOL(snd_ad1848_out);
static void snd_ad1848_dout(struct snd_ad1848 *chip,
unsigned char reg, unsigned char value)
{
- int timeout;
-
- for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--)
- udelay(100);
+ snd_ad1848_wait(chip);
outb(chip->mce_bit | reg, AD1848P(chip, REGSEL));
outb(value, AD1848P(chip, REG));
mb();
@@ -135,13 +140,11 @@ static void snd_ad1848_dout(struct snd_ad1848 *chip,
static unsigned char snd_ad1848_in(struct snd_ad1848 *chip, unsigned char reg)
{
- int timeout;
-
- for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--)
- udelay(100);
+ snd_ad1848_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT)
- snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x\n", reg);
+ snd_printk(KERN_WARNING "auto calibration time out - "
+ "reg = 0x%x\n", reg);
#endif
outb(chip->mce_bit | reg, AD1848P(chip, REGSEL));
mb();
@@ -183,8 +186,7 @@ static void snd_ad1848_mce_up(struct snd_ad1848 *chip)
unsigned long flags;
int timeout;
- for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--)
- udelay(100);
+ snd_ad1848_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT)
snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n");
@@ -201,9 +203,8 @@ static void snd_ad1848_mce_up(struct snd_ad1848 *chip)
static void snd_ad1848_mce_down(struct snd_ad1848 *chip)
{
- unsigned long flags;
- int timeout;
- signed long time;
+ unsigned long flags, timeout;
+ int reg;
spin_lock_irqsave(&chip->reg_lock, flags);
for (timeout = 5; timeout > 0; timeout--)
@@ -211,61 +212,48 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip)
/* end of cleanup sequence */
for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--)
udelay(100);
-#if 0
- printk("(1) timeout = %i\n", timeout);
-#endif
+
+ snd_printdd("(1) timeout = %d\n", timeout);
+
#ifdef CONFIG_SND_DEBUG
if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT)
snd_printk(KERN_WARNING "mce_down [0x%lx] - auto calibration time out (0)\n", AD1848P(chip, REGSEL));
#endif
+
chip->mce_bit &= ~AD1848_MCE;
- timeout = inb(AD1848P(chip, REGSEL));
- outb(chip->mce_bit | (timeout & 0x1f), AD1848P(chip, REGSEL));
- if (timeout == 0x80)
+ reg = inb(AD1848P(chip, REGSEL));
+ outb(chip->mce_bit | (reg & 0x1f), AD1848P(chip, REGSEL));
+ if (reg == 0x80)
snd_printk(KERN_WARNING "mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port);
- if ((timeout & AD1848_MCE) == 0) {
+ if ((reg & AD1848_MCE) == 0) {
spin_unlock_irqrestore(&chip->reg_lock, flags);
return;
}
- /* calibration process */
- for (timeout = 500; timeout > 0 && (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0; timeout--);
- if ((snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0) {
- snd_printd("mce_down - auto calibration time out (1)\n");
- spin_unlock_irqrestore(&chip->reg_lock, flags);
- return;
- }
-#if 0
- printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies);
-#endif
- time = HZ / 4;
- while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) {
+ /*
+ * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low.
+ * It may take up to 5 sample periods (at most 907 us @ 5.5125 kHz) for
+ * the process to _start_, so it is important to wait at least that long
+ * before checking. Otherwise we might think AC has finished when it
+ * has in fact not begun. It could take 128 (no AC) or 384 (AC) cycles
+ * for ACI to drop. This gives a wait of at most 70 ms with a more
+ * typical value of 3-9 ms.
+ */
+ timeout = jiffies + msecs_to_jiffies(250);
+ do {
spin_unlock_irqrestore(&chip->reg_lock, flags);
- if (time <= 0) {
- snd_printk(KERN_ERR "mce_down - auto calibration time out (2)\n");
- return;
- }
- time = schedule_timeout(time);
+ msleep(1);
spin_lock_irqsave(&chip->reg_lock, flags);
- }
-#if 0
- printk("(3) jiffies = %li\n", jiffies);
-#endif
- time = HZ / 10;
- while (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) {
- spin_unlock_irqrestore(&chip->reg_lock, flags);
- if (time <= 0) {
- snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n");
- return;
- }
- time = schedule_timeout(time);
- spin_lock_irqsave(&chip->reg_lock, flags);
- }
+ reg = snd_ad1848_in(chip, AD1848_TEST_INIT) &
+ AD1848_CALIB_IN_PROGRESS;
+ } while (reg && time_before(jiffies, timeout));
spin_unlock_irqrestore(&chip->reg_lock, flags);
-#if 0
- printk("(4) jiffies = %li\n", jiffies);
- snd_printk("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL)));
-#endif
+ if (reg)
+ snd_printk(KERN_ERR
+ "mce_down - auto calibration time out (2)\n");
+
+ snd_printdd("(4) jiffies = %lu\n", jiffies);
+ snd_printd("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL)));
}
static unsigned int snd_ad1848_get_count(unsigned char format,
@@ -319,11 +307,11 @@ static unsigned char snd_ad1848_get_rate(unsigned int rate)
{
int i;
- for (i = 0; i < 14; i++)
+ for (i = 0; i < ARRAY_SIZE(rates); i++)
if (rate == rates[i])
return freq_bits[i];
snd_BUG();
- return freq_bits[13];
+ return freq_bits[ARRAY_SIZE(rates) - 1];
}
static int snd_ad1848_ioctl(struct snd_pcm_substream *substream,
@@ -390,11 +378,9 @@ static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode)
{
unsigned long flags;
- mutex_lock(&chip->open_mutex);
- if (chip->mode & AD1848_MODE_OPEN) {
- mutex_unlock(&chip->open_mutex);
+ if (chip->mode & AD1848_MODE_OPEN)
return -EAGAIN;
- }
+
snd_ad1848_mce_down(chip);
#ifdef SNDRV_DEBUG_MCE
@@ -435,7 +421,6 @@ static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode)
spin_unlock_irqrestore(&chip->reg_lock, flags);
chip->mode = mode;
- mutex_unlock(&chip->open_mutex);
return 0;
}
@@ -444,11 +429,8 @@ static void snd_ad1848_close(struct snd_ad1848 *chip)
{
unsigned long flags;
- mutex_lock(&chip->open_mutex);
- if (!chip->mode) {
- mutex_unlock(&chip->open_mutex);
+ if (!chip->mode)
return;
- }
/* disable IRQ */
spin_lock_irqsave(&chip->reg_lock, flags);
outb(0, AD1848P(chip, STATUS)); /* clear IRQ */
@@ -474,7 +456,6 @@ static void snd_ad1848_close(struct snd_ad1848 *chip)
spin_unlock_irqrestore(&chip->reg_lock, flags);
chip->mode = 0;
- mutex_unlock(&chip->open_mutex);
}
/*
@@ -892,7 +873,6 @@ int snd_ad1848_create(struct snd_card *card,
if (chip == NULL)
return -ENOMEM;
spin_lock_init(&chip->reg_lock);
- mutex_init(&chip->open_mutex);
chip->card = card;
chip->port = port;
chip->irq = -1;
diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile
index 2fb4f7409d7c..5067ee001933 100644
--- a/sound/isa/cs423x/Makefile
+++ b/sound/isa/cs423x/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-cs4231-lib-objs := cs4231_lib.o
@@ -10,17 +10,8 @@ snd-cs4232-objs := cs4232.o
snd-cs4236-objs := cs4236.o
# Toplevel Module Dependency
-obj-$(CONFIG_SND_AZT2320) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_MIRO) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_OPL3SA2) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_CS4231) += snd-cs4231.o snd-cs4231-lib.o
-obj-$(CONFIG_SND_CS4232) += snd-cs4232.o snd-cs4231-lib.o
-obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o snd-cs4231-lib.o
-obj-$(CONFIG_SND_GUSMAX) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_INTERWAVE) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_INTERWAVE_STB) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_OPTI92X_CS4231) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_WAVEFRONT) += snd-cs4231-lib.o
-obj-$(CONFIG_SND_SSCAPE) += snd-cs4231-lib.o
+obj-$(CONFIG_SND_CS4231_LIB) += snd-cs4231-lib.o
+obj-$(CONFIG_SND_CS4231) += snd-cs4231.o
+obj-$(CONFIG_SND_CS4232) += snd-cs4232.o
+obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o
-obj-m := $(sort $(obj-m))
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index ac4041134150..13db6842eaaa 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -1,6 +1,6 @@
/*
* Generic driver for CS4231 chips
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Originally the CS4232/CS4232A driver, modified for use on CS4231 by
* Tugrul Galatali <galatalt@stuy.edu>
*
@@ -36,7 +36,7 @@
#define DEV_NAME "cs4231"
MODULE_DESCRIPTION(CRD_NAME);
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4231}}");
diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c
index 914d77b61b0c..a5eb9659b519 100644
--- a/sound/isa/cs423x/cs4231_lib.c
+++ b/sound/isa/cs423x/cs4231_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of CS4231(A)/CS4232/InterWave & compatible chips
*
* Bugs:
@@ -39,7 +39,7 @@
#include <asm/dma.h>
#include <asm/irq.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips");
MODULE_LICENSE("GPL");
@@ -74,7 +74,7 @@ static unsigned int rates[14] = {
};
static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
- .count = 14,
+ .count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
@@ -134,29 +134,31 @@ static inline u8 cs4231_inb(struct snd_cs4231 *chip, u8 offset)
return inb(chip->port + offset);
}
-static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg,
- unsigned char mask, unsigned char value)
+static void snd_cs4231_wait(struct snd_cs4231 *chip)
{
int timeout;
- unsigned char tmp;
for (timeout = 250;
timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT);
timeout--)
udelay(100);
+}
+
+static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg,
+ unsigned char mask, unsigned char value)
+{
+ unsigned char tmp = (chip->image[reg] & mask) | value;
+
+ snd_cs4231_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
#endif
- if (chip->calibrate_mute) {
- chip->image[reg] &= mask;
- chip->image[reg] |= value;
- } else {
+ chip->image[reg] = tmp;
+ if (!chip->calibrate_mute) {
cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
- mb();
- tmp = (chip->image[reg] & mask) | value;
+ wmb();
cs4231_outb(chip, CS4231P(REG), tmp);
- chip->image[reg] = tmp;
mb();
}
}
@@ -176,12 +178,7 @@ static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned
void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value)
{
- int timeout;
-
- for (timeout = 250;
- timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT);
- timeout--)
- udelay(100);
+ snd_cs4231_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
@@ -190,19 +187,13 @@ void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char va
cs4231_outb(chip, CS4231P(REG), value);
chip->image[reg] = value;
mb();
-#if 0
- printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value);
-#endif
+ snd_printdd("codec out - reg 0x%x = 0x%x\n",
+ chip->mce_bit | reg, value);
}
unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg)
{
- int timeout;
-
- for (timeout = 250;
- timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT);
- timeout--)
- udelay(100);
+ snd_cs4231_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
snd_printk("in: auto calibration time out - reg = 0x%x\n", reg);
@@ -304,8 +295,7 @@ void snd_cs4231_mce_up(struct snd_cs4231 *chip)
unsigned long flags;
int timeout;
- for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--)
- udelay(100);
+ snd_cs4231_wait(chip);
#ifdef CONFIG_SND_DEBUG
if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
snd_printk("mce_up - auto calibration time out (0)\n");
@@ -323,12 +313,11 @@ void snd_cs4231_mce_up(struct snd_cs4231 *chip)
void snd_cs4231_mce_down(struct snd_cs4231 *chip)
{
unsigned long flags;
+ unsigned long end_time;
int timeout;
snd_cs4231_busy_wait(chip);
-#if 0
- printk("(1) timeout = %i\n", timeout);
-#endif
+
#ifdef CONFIG_SND_DEBUG
if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL));
@@ -346,42 +335,42 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip)
}
snd_cs4231_busy_wait(chip);
- /* calibration process */
+ /*
+ * Wait for (possible -- during init auto-calibration may not be set)
+ * calibration process to start. Needs upto 5 sample periods on AD1848
+ * which at the slowest possible rate of 5.5125 kHz means 907 us.
+ */
+ msleep(1);
- for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--)
- udelay(10);
- if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) {
- snd_printd("cs4231_mce_down - auto calibration time out (1)\n");
- return;
- }
-#if 0
- printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies);
-#endif
- /* in 10 ms increments, check condition, up to 250 ms */
- timeout = 25;
- while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) {
- if (--timeout < 0) {
- snd_printk("mce_down - auto calibration time out (2)\n");
+ snd_printdd("(1) jiffies = %lu\n", jiffies);
+
+ /* check condition up to 250 ms */
+ end_time = jiffies + msecs_to_jiffies(250);
+ while (snd_cs4231_in(chip, CS4231_TEST_INIT) &
+ CS4231_CALIB_IN_PROGRESS) {
+
+ if (time_after(jiffies, end_time)) {
+ snd_printk(KERN_ERR "mce_down - "
+ "auto calibration time out (2)\n");
return;
}
- msleep(10);
+ msleep(1);
}
-#if 0
- printk("(3) jiffies = %li\n", jiffies);
-#endif
- /* in 10 ms increments, check condition, up to 100 ms */
- timeout = 10;
+
+ snd_printdd("(2) jiffies = %lu\n", jiffies);
+
+ /* check condition up to 100 ms */
+ end_time = jiffies + msecs_to_jiffies(100);
while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) {
- if (--timeout < 0) {
+ if (time_after(jiffies, end_time)) {
snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n");
return;
}
- msleep(10);
+ msleep(1);
}
-#if 0
- printk("(4) jiffies = %li\n", jiffies);
- snd_printk("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL)));
-#endif
+
+ snd_printdd("(3) jiffies = %lu\n", jiffies);
+ snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL)));
}
static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size)
@@ -459,11 +448,11 @@ static unsigned char snd_cs4231_get_rate(unsigned int rate)
{
int i;
- for (i = 0; i < 14; i++)
+ for (i = 0; i < ARRAY_SIZE(rates); i++)
if (rate == rates[i])
return freq_bits[i];
// snd_BUG();
- return freq_bits[13];
+ return freq_bits[ARRAY_SIZE(rates) - 1];
}
static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip,
@@ -555,6 +544,8 @@ static void snd_cs4231_playback_format(struct snd_cs4231 *chip,
snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr);
}
spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (chip->hardware == CS4231_HW_OPL3SA2)
+ udelay(100); /* this seems to help */
snd_cs4231_mce_down(chip);
}
snd_cs4231_calibrate_mute(chip, 0);
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 1a14f33b6ab0..5784b43f4123 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -1,6 +1,6 @@
/*
* Driver for generic CS4232/CS4235/CS4236/CS4236B/CS4237B/CS4238B/CS4239 chips
- * 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
@@ -32,7 +32,7 @@
#include <sound/opl3.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
#ifdef CS4232
MODULE_DESCRIPTION("Cirrus Logic CS4232");
diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c
index 7a5a6c71f5e4..6bd064470d4c 100644
--- a/sound/isa/cs423x/cs4236_lib.c
+++ b/sound/isa/cs423x/cs4236_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of CS4235/4236B/4237B/4238B/4239 chips
*
* Note:
@@ -89,7 +89,7 @@
#include <sound/cs4231.h>
#include <sound/asoundef.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips");
MODULE_LICENSE("GPL");
diff --git a/sound/isa/es1688/Makefile b/sound/isa/es1688/Makefile
index 501c8bf903af..aee1e4ddb22a 100644
--- a/sound/isa/es1688/Makefile
+++ b/sound/isa/es1688/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-es1688-lib-objs := es1688_lib.o
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index edc398712e8b..74bbc92f2e7c 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -1,6 +1,6 @@
/*
* Driver for generic ESS AudioDrive ESx688 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
@@ -39,7 +39,7 @@
#define DEV_NAME "es1688"
MODULE_DESCRIPTION(CRD_NAME);
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100},"
"{ESS,ES1688 PnP AudioDrive,pnp:ESS0102},"
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index a2ab99f2ac35..5c26d495daa8 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of ESS ES1688/688/488 chip
*
*
@@ -32,7 +32,7 @@
#include <asm/io.h>
#include <asm/dma.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("ESS ESx688 lowlevel module");
MODULE_LICENSE("GPL");
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index f7732bf90be3..4a7367a8ff9d 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1071,14 +1071,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
return (snd_es18xx_mixer_bits(chip, 0x1c, 0x07, val) != val) || retVal;
}
-static int snd_es18xx_info_spatializer_enable(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 snd_es18xx_info_spatializer_enable snd_ctl_boolean_mono_info
static int snd_es18xx_get_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1120,14 +1113,7 @@ static int snd_es18xx_get_hw_volume(struct snd_kcontrol *kcontrol, struct snd_ct
return 0;
}
-static int snd_es18xx_info_hw_switch(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 snd_es18xx_info_hw_switch snd_ctl_boolean_stereo_info
static int snd_es18xx_get_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2042,6 +2028,7 @@ static int pnpc_registered;
static struct pnp_device_id snd_audiodrive_pnpbiosids[] = {
{ .id = "ESS1869" },
+ { .id = "ESS1879" },
{ .id = "" } /* end */
};
diff --git a/sound/isa/gus/Makefile b/sound/isa/gus/Makefile
index bae5dbd6c8e5..df3d59f25f5e 100644
--- a/sound/isa/gus/Makefile
+++ b/sound/isa/gus/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-gus-lib-objs := gus_main.o \
diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c
index 44ee5d3674a1..fc905141e8a5 100644
--- a/sound/isa/gus/gus_dma.c
+++ b/sound/isa/gus/gus_dma.c
@@ -1,6 +1,6 @@
/*
* Routines for GF1 DMA control
- * 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
diff --git a/sound/isa/gus/gus_dram.c b/sound/isa/gus/gus_dram.c
index f22fe7967fcc..9eaa932f6efe 100644
--- a/sound/isa/gus/gus_dram.c
+++ b/sound/isa/gus/gus_dram.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* DRAM access routines
*
*
diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c
index d0c38e1856ef..bf137ea72329 100644
--- a/sound/isa/gus/gus_instr.c
+++ b/sound/isa/gus/gus_instr.c
@@ -1,6 +1,6 @@
/*
* Routines for Gravis UltraSound soundcards - Synthesizer
- * 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
diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c
index 9b1fe292de4d..3d4f899285ef 100644
--- a/sound/isa/gus/gus_io.c
+++ b/sound/isa/gus/gus_io.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* I/O routines for GF1/InterWave synthesizer chips
*
*
diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c
index 537d3cfe41f3..cd9a6f1c99e6 100644
--- a/sound/isa/gus/gus_irq.c
+++ b/sound/isa/gus/gus_irq.c
@@ -1,6 +1,6 @@
/*
* Routine for IRQ handling from GF1/InterWave chip
- * 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
@@ -45,11 +45,13 @@ __again:
// snd_printk("IRQ: status = 0x%x\n", status);
if (status & 0x02) {
STAT_ADD(gus->gf1.interrupt_stat_midi_in);
- gus->gf1.interrupt_handler_midi_in(gus);
+ if (gus->gf1.interrupt_handler_midi_in)
+ gus->gf1.interrupt_handler_midi_in(gus);
}
if (status & 0x01) {
STAT_ADD(gus->gf1.interrupt_stat_midi_out);
- gus->gf1.interrupt_handler_midi_out(gus);
+ if (gus->gf1.interrupt_handler_midi_out)
+ gus->gf1.interrupt_handler_midi_out(gus);
}
if (status & (0x20 | 0x40)) {
unsigned int already, _current_;
@@ -85,20 +87,24 @@ __again:
}
if (status & 0x04) {
STAT_ADD(gus->gf1.interrupt_stat_timer1);
- gus->gf1.interrupt_handler_timer1(gus);
+ if (gus->gf1.interrupt_handler_timer1)
+ gus->gf1.interrupt_handler_timer1(gus);
}
if (status & 0x08) {
STAT_ADD(gus->gf1.interrupt_stat_timer2);
- gus->gf1.interrupt_handler_timer2(gus);
+ if (gus->gf1.interrupt_handler_timer2)
+ gus->gf1.interrupt_handler_timer2(gus);
}
if (status & 0x80) {
if (snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL) & 0x40) {
STAT_ADD(gus->gf1.interrupt_stat_dma_write);
- gus->gf1.interrupt_handler_dma_write(gus);
+ if (gus->gf1.interrupt_handler_dma_write)
+ gus->gf1.interrupt_handler_dma_write(gus);
}
if (snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL) & 0x40) {
STAT_ADD(gus->gf1.interrupt_stat_dma_read);
- gus->gf1.interrupt_handler_dma_read(gus);
+ if (gus->gf1.interrupt_handler_dma_read)
+ gus->gf1.interrupt_handler_dma_read(gus);
}
}
if (--loop > 0)
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index 8ced5e81b9a7..b14d5d6d9a32 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -1,6 +1,6 @@
/*
* Routines for Gravis UltraSound 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
@@ -31,7 +31,7 @@
#include <asm/dma.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards");
MODULE_LICENSE("GPL");
@@ -154,6 +154,14 @@ int snd_gus_create(struct snd_card *card,
gus = kzalloc(sizeof(*gus), GFP_KERNEL);
if (gus == NULL)
return -ENOMEM;
+ spin_lock_init(&gus->reg_lock);
+ spin_lock_init(&gus->voice_alloc);
+ spin_lock_init(&gus->active_voice_lock);
+ spin_lock_init(&gus->event_lock);
+ spin_lock_init(&gus->dma_lock);
+ spin_lock_init(&gus->pcm_volume_level_lock);
+ spin_lock_init(&gus->uart_cmd_lock);
+ mutex_init(&gus->dma_mutex);
gus->gf1.irq = -1;
gus->gf1.dma1 = -1;
gus->gf1.dma2 = -1;
@@ -218,14 +226,6 @@ int snd_gus_create(struct snd_card *card,
gus->gf1.pcm_channels = pcm_channels;
gus->gf1.volume_ramp = 25;
gus->gf1.smooth_pan = 1;
- spin_lock_init(&gus->reg_lock);
- spin_lock_init(&gus->voice_alloc);
- spin_lock_init(&gus->active_voice_lock);
- spin_lock_init(&gus->event_lock);
- spin_lock_init(&gus->dma_lock);
- spin_lock_init(&gus->pcm_volume_level_lock);
- spin_lock_init(&gus->uart_cmd_lock);
- mutex_init(&gus->dma_mutex);
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, gus, &ops)) < 0) {
snd_gus_free(gus);
return err;
@@ -398,7 +398,7 @@ static int snd_gus_check_version(struct snd_gus_card * gus)
gus->ess_flag = 1;
} else {
snd_printk(KERN_ERR "unknown GF1 revision number at 0x%lx - 0x%x (0x%x)\n", gus->gf1.port, rev, val);
- snd_printk(KERN_ERR " please - report to <perex@suse.cz>\n");
+ snd_printk(KERN_ERR " please - report to <perex@perex.cz>\n");
}
}
}
diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c
index 7107753b85b5..bcf4656853c4 100644
--- a/sound/isa/gus/gus_mem.c
+++ b/sound/isa/gus/gus_mem.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* GUS's memory allocation routines / bottom layer
*
*
diff --git a/sound/isa/gus/gus_mem_proc.c b/sound/isa/gus/gus_mem_proc.c
index 80f0a83818b2..f69a44728ebf 100644
--- a/sound/isa/gus/gus_mem_proc.c
+++ b/sound/isa/gus/gus_mem_proc.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* GUS's memory access via proc filesystem
*
*
diff --git a/sound/isa/gus/gus_mixer.c b/sound/isa/gus/gus_mixer.c
index acc25a297200..a96253e16654 100644
--- a/sound/isa/gus/gus_mixer.c
+++ b/sound/isa/gus/gus_mixer.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of ICS 2101 chip and "mixer" in GF1 chip
*
*
@@ -36,14 +36,7 @@
.get = snd_gf1_get_single, .put = snd_gf1_put_single, \
.private_value = shift | (invert << 8) }
-static int snd_gf1_info_single(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 snd_gf1_info_single snd_ctl_boolean_mono_info
static int snd_gf1_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index c7f95e7aa018..a7971f5ffe63 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of GF1 chip (PCM things)
*
* InterWave chips supports interleaved DMA, but this feature isn't used in
diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c
index b263655c4116..20cfdb87f84a 100644
--- a/sound/isa/gus/gus_reset.c
+++ b/sound/isa/gus/gus_reset.c
@@ -1,5 +1,5 @@
/*
- * 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
diff --git a/sound/isa/gus/gus_sample.c b/sound/isa/gus/gus_sample.c
index 9e0c55ab25b2..cba0829a7106 100644
--- a/sound/isa/gus/gus_sample.c
+++ b/sound/isa/gus/gus_sample.c
@@ -1,6 +1,6 @@
/*
* Routines for Gravis UltraSound soundcards - Sample 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
diff --git a/sound/isa/gus/gus_simple.c b/sound/isa/gus/gus_simple.c
index dcad6ed0198c..39d121e2c8c4 100644
--- a/sound/isa/gus/gus_simple.c
+++ b/sound/isa/gus/gus_simple.c
@@ -1,6 +1,6 @@
/*
* Routines for Gravis UltraSound soundcards - Simple instrument handlers
- * 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
diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c
index 3e4d4d6edd8b..2c2051782aa2 100644
--- a/sound/isa/gus/gus_synth.c
+++ b/sound/isa/gus/gus_synth.c
@@ -1,6 +1,6 @@
/*
* Routines for Gravis UltraSound soundcards - Synthesizer
- * 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
@@ -26,7 +26,7 @@
#include <sound/gus.h>
#include <sound/seq_device.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards - Synthesizer");
MODULE_LICENSE("GPL");
diff --git a/sound/isa/gus/gus_tables.h b/sound/isa/gus/gus_tables.h
index 4adf098d3269..42a4ca0d622b 100644
--- a/sound/isa/gus/gus_tables.h
+++ b/sound/isa/gus/gus_tables.h
@@ -1,5 +1,5 @@
/*
- * 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
diff --git a/sound/isa/gus/gus_timer.c b/sound/isa/gus/gus_timer.c
index a43b662f17c7..99eac573c414 100644
--- a/sound/isa/gus/gus_timer.c
+++ b/sound/isa/gus/gus_timer.c
@@ -1,6 +1,6 @@
/*
* Routines for Gravis UltraSound soundcards - Timers
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* GUS have similar timers as AdLib (OPL2/OPL3 chips).
*
diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c
index 654290a8b21c..e6fd9b01c492 100644
--- a/sound/isa/gus/gus_uart.c
+++ b/sound/isa/gus/gus_uart.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for the GF1 MIDI interface - like UART 6850
*
*
diff --git a/sound/isa/gus/gus_volume.c b/sound/isa/gus/gus_volume.c
index dbbc0a6d7659..71a67744a14b 100644
--- a/sound/isa/gus/gus_volume.c
+++ b/sound/isa/gus/gus_volume.c
@@ -1,5 +1,5 @@
/*
- * 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
diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c
index 8f23f433d491..29e422b00b58 100644
--- a/sound/isa/gus/gusclassic.c
+++ b/sound/isa/gus/gusclassic.c
@@ -1,6 +1,6 @@
/*
* Driver for Gravis UltraSound Classic soundcard
- * 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
@@ -37,7 +37,7 @@
#define DEV_NAME "gusclassic"
MODULE_DESCRIPTION(CRD_NAME);
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Classic}}");
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index 0aeaa6cf6cf0..fc59536c918e 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -1,6 +1,6 @@
/*
* Driver for Gravis UltraSound Extreme 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
@@ -41,7 +41,7 @@
#define DEV_NAME "gusextreme"
MODULE_DESCRIPTION(CRD_NAME);
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Extreme}}");
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
index 708783d4351f..4922f5da08f9 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -1,6 +1,6 @@
/*
* Driver for Gravis UltraSound MAX soundcard
- * 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
@@ -34,7 +34,7 @@
#define SNDRV_LEGACY_FIND_FREE_DMA
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Gravis UltraSound MAX");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound MAX}}");
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 0220cdbe1a2a..2091c50b2e3e 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -1,6 +1,6 @@
/*
* Driver for AMD InterWave soundcard
- * 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
@@ -41,7 +41,7 @@
#define SNDRV_LEGACY_FIND_FREE_DMA
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
#ifndef SNDRV_STB
MODULE_DESCRIPTION("AMD InterWave");
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index e70db32991d9..59af9ab7191f 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -1,6 +1,6 @@
/*
* Driver for Yamaha OPL3-SA[2,3] 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
@@ -37,7 +37,7 @@
#include <asm/io.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Yamaha OPL3SA2+");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719E-S},"
@@ -253,6 +253,7 @@ static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip)
/* 0x03 - YM715B */
/* 0x04 - YM719 - OPL-SA4? */
/* 0x05 - OPL3-SA3 - Libretto 100 */
+ /* 0x07 - unknown - Neomagic MagicWave 3D */
break;
}
str[0] = chip->version + '0';
diff --git a/sound/isa/opti9xx/Makefile b/sound/isa/opti9xx/Makefile
index 0e41bfd5a403..b4d894db257a 100644
--- a/sound/isa/opti9xx/Makefile
+++ b/sound/isa/opti9xx/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-opti92x-ad1848-objs := opti92x-ad1848.o
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index cd29b30b362e..d295936611f8 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -242,14 +242,7 @@ static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value)
* MIXER part
*/
-static int snd_miro_info_capture(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
-
- return 0;
-}
+#define snd_miro_info_capture snd_ctl_boolean_mono_info
static int snd_miro_get_capture(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -344,14 +337,7 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol,
return change;
}
-static int snd_miro_info_amp(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
-
- return 0;
-}
+#define snd_miro_info_amp snd_ctl_boolean_mono_info
static int snd_miro_get_amp(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 049d479ce2b3..ee1a824d8fc0 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -501,6 +501,16 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip)
(chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04),
0x34);
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf);
+ /*
+ * The BTC 1817DW has QS1000 wavetable which is connected
+ * to the serial digital input of the OPTI931.
+ */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(21), 0x82, 0xff);
+ /*
+ * This bit sets OPTI931 to automaticaly select FM
+ * or digital input signal.
+ */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
break;
#endif /* OPTi93X */
@@ -1732,11 +1742,11 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip,
#ifdef OPTi93X
port = pnp_port_start(pdev, 0) - 4;
- fm_port = pnp_port_start(pdev, 1);
+ fm_port = pnp_port_start(pdev, 1) + 8;
#else
if (pid->driver_data != 0x0924)
port = pnp_port_start(pdev, 1);
- fm_port = pnp_port_start(pdev, 2);
+ fm_port = pnp_port_start(pdev, 2) + 8;
#endif /* OPTi93X */
irq = pnp_irq(pdev, 0);
dma1 = pnp_dma(pdev, 0);
diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile
index 556e66928029..c9d1c986d70e 100644
--- a/sound/isa/sb/Makefile
+++ b/sound/isa/sb/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-sb-common-objs := sb_common.o sb_mixer.o
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 658179e86142..4eea84cfd4f4 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* and (c) 1999 Steve Ratcliffe <steve@parabola.demon.co.uk>
* Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
*
diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c
index 3d72742b342f..0c7905c85b76 100644
--- a/sound/isa/sb/emu8000_synth.c
+++ b/sound/isa/sb/emu8000_synth.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* and (c) 1999 Steve Ratcliffe <steve@parabola.demon.co.uk>
* Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
*
diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c
index c4ba24bfd27c..e7f9edd92626 100644
--- a/sound/isa/sb/sb16.c
+++ b/sound/isa/sb/sb16.c
@@ -1,6 +1,6 @@
/*
* Driver for SoundBlaster 16/AWE32/AWE64 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
@@ -44,7 +44,7 @@
#define PFX "sb16: "
#endif
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_LICENSE("GPL");
#ifndef SNDRV_SBAWE
MODULE_DESCRIPTION("Sound Blaster 16");
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index b279f2308aef..3682059787ab 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -979,14 +979,7 @@ static int snd_sb_csp_restart(struct snd_sb_csp * p)
* QSound mixer control for PCM
*/
-static int snd_sb_qsound_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 snd_sb_qsound_switch_info snd_ctl_boolean_mono_info
static int snd_sb_qsound_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c
index 5d4d3aafe2d5..c06754f7ee5d 100644
--- a/sound/isa/sb/sb16_main.c
+++ b/sound/isa/sb/sb16_main.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of 16-bit SoundBlaster cards and clones
* Note: This is very ugly hardware which uses one 8-bit DMA channel and
* second 16-bit DMA channel. Unfortunately 8-bit DMA channel can't
@@ -45,7 +45,7 @@
#include <sound/control.h>
#include <sound/info.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones");
MODULE_LICENSE("GPL");
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index a1b3786b391e..f933aef7d8a9 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -1,6 +1,6 @@
/*
* Driver for SoundBlaster 1.0/2.0/Pro soundcards and 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
@@ -31,7 +31,7 @@
#include <sound/opl3.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Sound Blaster 1.0/2.0/Pro");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 1.0/SB 2.0/SB Pro}}");
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index aea9e5ec7b36..bee894b3f5c7 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Uros Bizjak <uros@kss-loka.si>
*
* Routines for control of 8-bit SoundBlaster cards and clones
@@ -38,7 +38,7 @@
#include <sound/core.h>
#include <sound/sb.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Uros Bizjak <uros@kss-loka.si>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Uros Bizjak <uros@kss-loka.si>");
MODULE_DESCRIPTION("Routines for control of 8-bit SoundBlaster cards and clones");
MODULE_LICENSE("GPL");
diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c
index 0b67edd7ac6e..e56e5633411c 100644
--- a/sound/isa/sb/sb8_midi.c
+++ b/sound/isa/sb/sb8_midi.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of SoundBlaster cards - MIDI interface
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index efa9d5c2558a..176193c05101 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Uros Bizjak <uros@kss-loka.si>
*
* Lowlevel routines for control of Sound Blaster cards
@@ -33,7 +33,7 @@
#include <asm/io.h>
#include <asm/dma.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("ALSA lowlevel driver for Sound Blaster cards");
MODULE_LICENSE("GPL");
@@ -234,7 +234,9 @@ int snd_sbdsp_create(struct snd_card *card,
chip->dma16 = -1;
chip->port = port;
- if (request_irq(irq, irq_handler, hardware == SB_HW_ALS4000 ?
+ if (request_irq(irq, irq_handler,
+ (hardware == SB_HW_ALS4000 ||
+ hardware == SB_HW_CS5530) ?
IRQF_SHARED : IRQF_DISABLED,
"SoundBlaster", (void *) chip)) {
snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq);
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 3d4befcff28e..03241cd5aaef 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for Sound Blaster mixer control
*
*
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
new file mode 100644
index 000000000000..94daf8399994
--- /dev/null
+++ b/sound/isa/sc6000.c
@@ -0,0 +1,656 @@
+/*
+ * Driver for Gallant SC-6000 soundcard. This card is also known as
+ * Audio Excel DSP 16 or Zoltrix AV302.
+ * These cards use CompuMedia ASC-9308 chip + AD1848 codec.
+ *
+ * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1@wp.pl>
+ *
+ * I don't have documentation for this card. I used the driver
+ * for OSS/Free included in the kernel source as reference.
+ *
+ * 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
+ */
+
+#include <sound/driver.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/isa.h>
+#include <linux/io.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/ad1848.h>
+#include <sound/opl3.h>
+#include <sound/mpu401.h>
+#include <sound/control.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Krzysztof Helt");
+MODULE_DESCRIPTION("Gallant SC-6000");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000},"
+ "{AudioExcel, Audio Excel DSP 16},"
+ "{Zoltrix, AV302}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */
+static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+ /* 0x300, 0x310, 0x320, 0x330 */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */
+static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for sc-6000 driver.");
+module_param_array(mss_port, long, NULL, 0444);
+MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver.");
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver.");
+module_param_array(dma, int, NULL, 0444);
+MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver.");
+
+/*
+ * Commands of SC6000's DSP (SBPRO+special).
+ * Some of them are COMMAND_xx, in the future they may change.
+ */
+#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */
+#define COMMAND_52 0x52 /* */
+#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */
+#define COMMAND_5C 0x5c /* */
+#define COMMAND_60 0x60 /* */
+#define COMMAND_66 0x66 /* */
+#define COMMAND_6C 0x6c /* */
+#define COMMAND_6E 0x6e /* */
+#define COMMAND_88 0x88 /* Unknown command */
+#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */
+#define COMMAND_C5 0xc5 /* */
+#define GET_DSP_VERSION 0xe1 /* Get DSP Version */
+#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */
+
+/*
+ * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port
+ * to have the actual I/O port.
+ * Register permissions are:
+ * (wo) == Write Only
+ * (ro) == Read Only
+ * (w-) == Write
+ * (r-) == Read
+ */
+#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */
+#define DSP_READ 0x0a /* offset of DSP READ (ro) */
+#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */
+#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */
+#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */
+#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */
+
+#define PFX "sc6000: "
+#define DRV_NAME "SC-6000"
+
+/* hardware dependent functions */
+
+/*
+ * sc6000_irq_to_softcfg - Decode irq number into cfg code.
+ */
+static __devinit unsigned char sc6000_irq_to_softcfg(int irq)
+{
+ unsigned char val = 0;
+
+ switch (irq) {
+ case 5:
+ val = 0x28;
+ break;
+ case 7:
+ val = 0x8;
+ break;
+ case 9:
+ val = 0x10;
+ break;
+ case 10:
+ val = 0x18;
+ break;
+ case 11:
+ val = 0x20;
+ break;
+ default:
+ break;
+ }
+ return val;
+}
+
+/*
+ * sc6000_dma_to_softcfg - Decode dma number into cfg code.
+ */
+static __devinit unsigned char sc6000_dma_to_softcfg(int dma)
+{
+ unsigned char val = 0;
+
+ switch (dma) {
+ case 0:
+ val = 1;
+ break;
+ case 1:
+ val = 2;
+ break;
+ case 3:
+ val = 3;
+ break;
+ default:
+ break;
+ }
+ return val;
+}
+
+/*
+ * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code.
+ */
+static __devinit unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq)
+{
+ unsigned char val = 0;
+
+ switch (mpu_irq) {
+ case 5:
+ val = 4;
+ break;
+ case 7:
+ val = 0x44;
+ break;
+ case 9:
+ val = 0x84;
+ break;
+ case 10:
+ val = 0xc4;
+ break;
+ default:
+ break;
+ }
+ return val;
+}
+
+static __devinit int sc6000_wait_data(char __iomem *vport)
+{
+ int loop = 1000;
+ unsigned char val = 0;
+
+ do {
+ val = ioread8(vport + DSP_DATAVAIL);
+ if (val & 0x80)
+ return 0;
+ cpu_relax();
+ } while (loop--);
+
+ return -EAGAIN;
+}
+
+static __devinit int sc6000_read(char __iomem *vport)
+{
+ if (sc6000_wait_data(vport))
+ return -EBUSY;
+
+ return ioread8(vport + DSP_READ);
+
+}
+
+static __devinit int sc6000_write(char __iomem *vport, int cmd)
+{
+ unsigned char val;
+ int loop = 500000;
+
+ do {
+ val = ioread8(vport + DSP_STATUS);
+ /*
+ * DSP ready to receive data if bit 7 of val == 0
+ */
+ if (!(val & 0x80)) {
+ iowrite8(cmd, vport + DSP_COMMAND);
+ return 0;
+ }
+ cpu_relax();
+ } while (loop--);
+
+ snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd);
+
+ return -EIO;
+}
+
+static int __devinit sc6000_dsp_get_answer(char __iomem *vport, int command,
+ char *data, int data_len)
+{
+ int len = 0;
+
+ if (sc6000_write(vport, command)) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command);
+ return -EIO;
+ }
+
+ do {
+ int val = sc6000_read(vport);
+
+ if (val < 0)
+ break;
+
+ data[len++] = val;
+
+ } while (len < data_len);
+
+ /*
+ * If no more data available, return to the caller, no error if len>0.
+ * We have no other way to know when the string is finished.
+ */
+ return len ? len : -EIO;
+}
+
+static int __devinit sc6000_dsp_reset(char __iomem *vport)
+{
+ iowrite8(1, vport + DSP_RESET);
+ udelay(10);
+ iowrite8(0, vport + DSP_RESET);
+ udelay(20);
+ if (sc6000_read(vport) == 0xaa)
+ return 0;
+ return -ENODEV;
+}
+
+/* detection and initialization */
+static int __devinit sc6000_cfg_write(char __iomem *vport,
+ unsigned char softcfg)
+{
+
+ if (sc6000_write(vport, WRITE_MDIRQ_CFG)) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG);
+ return -EIO;
+ }
+ if (sc6000_write(vport, softcfg)) {
+ snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n");
+ return -EIO;
+ }
+ return 0;
+}
+
+static int __devinit sc6000_setup_board(char __iomem *vport, int config)
+{
+ int loop = 10;
+
+ do {
+ if (sc6000_write(vport, COMMAND_88)) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n",
+ COMMAND_88);
+ return -EIO;
+ }
+ } while ((sc6000_wait_data(vport) < 0) && loop--);
+
+ if (sc6000_read(vport) < 0) {
+ snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n",
+ COMMAND_88);
+ return -EIO;
+ }
+
+ if (sc6000_cfg_write(vport, config))
+ return -ENODEV;
+
+ return 0;
+}
+
+static int __devinit sc6000_init_mss(char __iomem *vport, int config,
+ char __iomem *vmss_port, int mss_config)
+{
+ if (sc6000_write(vport, DSP_INIT_MSS)) {
+ snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n",
+ DSP_INIT_MSS);
+ return -EIO;
+ }
+
+ msleep(10);
+
+ if (sc6000_cfg_write(vport, config))
+ return -EIO;
+
+ iowrite8(mss_config, vmss_port);
+
+ return 0;
+}
+
+static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma,
+ char __iomem *vmss_port, int mpu_irq)
+{
+ char answer[15];
+ char version[2];
+ int mss_config = sc6000_irq_to_softcfg(irq) |
+ sc6000_dma_to_softcfg(dma);
+ int config = mss_config |
+ sc6000_mpu_irq_to_softcfg(mpu_irq);
+ int err;
+
+ err = sc6000_dsp_reset(vport);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n");
+ return err;
+ }
+
+ memset(answer, 0, sizeof(answer));
+ err = sc6000_dsp_get_answer(vport, GET_DSP_COPYRIGHT, answer, 15);
+ if (err <= 0) {
+ snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n");
+ return -ENODEV;
+ }
+ /*
+ * My SC-6000 card return "SC-6000" in DSPCopyright, so
+ * if we have something different, we have to be warned.
+ * Mine returns "SC-6000A " - KH
+ */
+ if (strncmp("SC-6000", answer, 7))
+ snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n");
+
+ if (sc6000_dsp_get_answer(vport, GET_DSP_VERSION, version, 2) < 2) {
+ snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n",
+ answer, version[0], version[1]);
+
+ /*
+ * 0x0A == (IRQ 7, DMA 1, MIRQ 0)
+ */
+ err = sc6000_cfg_write(vport, 0x0a);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n");
+ return -EFAULT;
+ }
+
+ err = sc6000_setup_board(vport, config);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sc6000_setup_board: failed!\n");
+ return -ENODEV;
+ }
+
+ err = sc6000_init_mss(vport, config, vmss_port, mss_config);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Can not initialize"
+ "Microsoft Sound System mode.\n");
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id1, id2;
+ int err;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ /* reassign AUX0 to FM */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "FM Playback Switch");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "FM Playback Volume");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ /* reassign AUX1 to CD */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "CD Playback Switch");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "CD Playback Volume");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+ if (port[dev] == SNDRV_AUTO_PORT) {
+ printk(KERN_ERR PFX "specify IO port\n");
+ return 0;
+ }
+ if (mss_port[dev] == SNDRV_AUTO_PORT) {
+ printk(KERN_ERR PFX "specify MSS port\n");
+ return 0;
+ }
+ if (port[dev] != 0x220 && port[dev] != 0x240) {
+ printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n");
+ return 0;
+ }
+ if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) {
+ printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n");
+ return 0;
+ }
+ if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) {
+ printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]);
+ return 0;
+ }
+ if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) {
+ printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]);
+ return 0;
+ }
+ if (mpu_port[dev] != SNDRV_AUTO_PORT &&
+ (mpu_port[dev] & ~0x30L) != 0x300) {
+ printk(KERN_ERR PFX "invalid MPU-401 port %lx\n",
+ mpu_port[dev]);
+ return 0;
+ }
+ if (mpu_port[dev] != SNDRV_AUTO_PORT &&
+ mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 &&
+ !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) {
+ printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]);
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev)
+{
+ static int possible_irqs[] = { 5, 7, 9, 10, 11, -1 };
+ static int possible_dmas[] = { 1, 3, 0, -1 };
+ int err;
+ int xirq = irq[dev];
+ int xdma = dma[dev];
+ struct snd_card *card;
+ struct snd_ad1848 *chip;
+ struct snd_opl3 *opl3;
+ char __iomem *vport;
+ char __iomem *vmss_port;
+
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (!card)
+ return -ENOMEM;
+
+ if (xirq == SNDRV_AUTO_IRQ) {
+ xirq = snd_legacy_find_free_irq(possible_irqs);
+ if (xirq < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ err = -EBUSY;
+ goto err_exit;
+ }
+ }
+
+ if (xdma == SNDRV_AUTO_DMA) {
+ xdma = snd_legacy_find_free_dma(possible_dmas);
+ if (xdma < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA\n");
+ err = -EBUSY;
+ goto err_exit;
+ }
+ }
+
+ if (!request_region(port[dev], 0x10, DRV_NAME)) {
+ snd_printk(KERN_ERR PFX
+ "I/O port region is already in use.\n");
+ err = -EBUSY;
+ goto err_exit;
+ }
+ vport = devm_ioport_map(devptr, port[dev], 0x10);
+ if (!vport) {
+ snd_printk(KERN_ERR PFX
+ "I/O port cannot be iomaped.\n");
+ err = -EBUSY;
+ goto err_unmap1;
+ }
+
+ /* to make it marked as used */
+ if (!request_region(mss_port[dev], 4, DRV_NAME)) {
+ snd_printk(KERN_ERR PFX
+ "SC-6000 port I/O port region is already in use.\n");
+ err = -EBUSY;
+ goto err_unmap1;
+ }
+ vmss_port = devm_ioport_map(devptr, mss_port[dev], 4);
+ if (!vport) {
+ snd_printk(KERN_ERR PFX
+ "MSS port I/O cannot be iomaped.\n");
+ err = -EBUSY;
+ goto err_unmap2;
+ }
+
+ snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n",
+ port[dev], xirq, xdma,
+ mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]);
+
+ err = sc6000_init_board(vport, xirq, xdma, vmss_port, mpu_irq[dev]);
+ if (err < 0)
+ goto err_unmap2;
+
+ err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma,
+ AD1848_HW_DETECT, &chip);
+ if (err < 0)
+ goto err_unmap2;
+ card->private_data = chip;
+
+ err = snd_ad1848_pcm(chip, 0, NULL);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX
+ "error creating new ad1848 PCM device\n");
+ goto err_unmap2;
+ }
+ err = snd_ad1848_mixer(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "error creating new ad1848 mixer\n");
+ goto err_unmap2;
+ }
+ err = snd_sc6000_mixer(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "the mixer rewrite failed\n");
+ goto err_unmap2;
+ }
+ if (snd_opl3_create(card,
+ 0x388, 0x388 + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n",
+ 0x388, 0x388 + 2);
+ } else {
+ err = snd_opl3_timer_new(opl3, 0, 1);
+ if (err < 0)
+ goto err_unmap2;
+
+ err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+ if (err < 0)
+ goto err_unmap2;
+ }
+
+ if (mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if (mpu_irq[dev] == SNDRV_AUTO_IRQ)
+ mpu_irq[dev] = -1;
+ if (snd_mpu401_uart_new(card, 0,
+ MPU401_HW_MPU401,
+ mpu_port[dev], 0,
+ mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
+ snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n",
+ mpu_port[dev]);
+ }
+
+ strcpy(card->driver, DRV_NAME);
+ strcpy(card->shortname, "SC-6000");
+ sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d",
+ mss_port[dev], xirq, xdma);
+
+ snd_card_set_dev(card, devptr);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto err_unmap2;
+
+ dev_set_drvdata(devptr, card);
+ return 0;
+
+err_unmap2:
+ release_region(mss_port[dev], 4);
+err_unmap1:
+ release_region(port[dev], 0x10);
+err_exit:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev)
+{
+ release_region(port[dev], 0x10);
+ release_region(mss_port[dev], 4);
+
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_sc6000_driver = {
+ .match = snd_sc6000_match,
+ .probe = snd_sc6000_probe,
+ .remove = __devexit_p(snd_sc6000_remove),
+ /* FIXME: suspend/resume */
+ .driver = {
+ .name = DRV_NAME,
+ },
+};
+
+
+static int __init alsa_card_sc6000_init(void)
+{
+ return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_sc6000_exit(void)
+{
+ isa_unregister_driver(&snd_sc6000_driver);
+}
+
+module_init(alsa_card_sc6000_init)
+module_exit(alsa_card_sc6000_exit)
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index cbad2a51cbaa..1cb921d6137e 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -45,10 +45,12 @@ MODULE_LICENSE("GPL");
static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX;
static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR;
-static long port[SNDRV_CARDS] __devinitdata = { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_PORT };
+static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT;
+static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT;
static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ;
static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ;
static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA;
+static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA;
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index number for SoundScape soundcard");
@@ -59,6 +61,9 @@ MODULE_PARM_DESC(id, "Description for SoundScape card");
module_param_array(port, long, NULL, 0444);
MODULE_PARM_DESC(port, "Port # for SoundScape driver.");
+module_param_array(wss_port, long, NULL, 0444);
+MODULE_PARM_DESC(wss_port, "WSS Port # for SoundScape driver.");
+
module_param_array(irq, int, NULL, 0444);
MODULE_PARM_DESC(irq, "IRQ # for SoundScape driver.");
@@ -68,12 +73,16 @@ MODULE_PARM_DESC(mpu_irq, "MPU401 IRQ # for SoundScape driver.");
module_param_array(dma, int, NULL, 0444);
MODULE_PARM_DESC(dma, "DMA # for SoundScape driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver.");
+
#ifdef CONFIG_PNP
static int isa_registered;
static int pnp_registered;
static struct pnp_card_device_id sscape_pnpids[] = {
- { .id = "ENS3081", .devs = { { "ENS0000" } } },
+ { .id = "ENS3081", .devs = { { "ENS0000" } } }, /* Soundscape PnP */
+ { .id = "ENS4081", .devs = { { "ENS1011" } } }, /* VIVO90 */
{ .id = "" } /* end */
};
@@ -124,12 +133,21 @@ enum GA_REG {
#define AD1845_FREQ_SEL_MSB 0x16
#define AD1845_FREQ_SEL_LSB 0x17
+enum card_type {
+ SSCAPE,
+ SSCAPE_PNP,
+ SSCAPE_VIVO,
+};
+
struct soundscape {
spinlock_t lock;
unsigned io_base;
+ unsigned wss_base;
int codec_type;
int ic_type;
+ enum card_type type;
struct resource *io_res;
+ struct resource *wss_res;
struct snd_cs4231 *chip;
struct snd_mpu401 *mpu;
struct snd_hwdep *hw;
@@ -340,8 +358,9 @@ static inline void activate_ad1845_unsafe(unsigned io_base)
*/
static void soundscape_free(struct snd_card *c)
{
- register struct soundscape *sscape = get_card_soundscape(c);
+ struct soundscape *sscape = get_card_soundscape(c);
release_and_free_resource(sscape->io_res);
+ release_and_free_resource(sscape->wss_res);
free_dma(sscape->chip->dma1);
}
@@ -382,7 +401,7 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout)
unsigned long flags;
unsigned char x;
- schedule_timeout(1);
+ schedule_timeout_uninterruptible(1);
spin_lock_irqsave(&s->lock, flags);
x = inb(HOST_DATA_IO(s->io_base));
@@ -409,7 +428,7 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout)
unsigned long flags;
unsigned char x;
- schedule_timeout(1);
+ schedule_timeout_uninterruptible(1);
spin_lock_irqsave(&s->lock, flags);
x = inb(HOST_DATA_IO(s->io_base));
@@ -522,7 +541,7 @@ static int upload_dma_data(struct soundscape *s,
ret = -EAGAIN;
}
- _release_dma:
+_release_dma:
/*
* NOTE!!! We are NOT holding any spinlocks at this point !!!
*/
@@ -802,6 +821,7 @@ static int __devinit detect_sscape(struct soundscape *s)
unsigned long flags;
unsigned d;
int retval = 0;
+ int codec = s->wss_base;
spin_lock_irqsave(&s->lock, flags);
@@ -833,9 +853,27 @@ static int __devinit detect_sscape(struct soundscape *s)
outb(0xfe, ODIE_ADDR_IO(s->io_base));
if ((inb(ODIE_ADDR_IO(s->io_base)) & 0x9f) != 0x0e)
goto _done;
- if ((inb(ODIE_DATA_IO(s->io_base)) & 0x9f) != 0x0e)
+
+ outb(0xfe, ODIE_ADDR_IO(s->io_base));
+ d = inb(ODIE_DATA_IO(s->io_base));
+ if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e)
goto _done;
+ d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f;
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0);
+
+ if (s->type == SSCAPE_VIVO)
+ codec += 4;
+ /* wait for WSS codec */
+ for (d = 0; d < 500; d++) {
+ if ((inb(codec) & 0x80) == 0)
+ break;
+ spin_unlock_irqrestore(&s->lock, flags);
+ msleep(1);
+ spin_lock_irqsave(&s->lock, flags);
+ }
+ snd_printd(KERN_INFO "init delay = %d ms\n", d);
+
/*
* SoundScape successfully detected!
*/
@@ -995,21 +1033,23 @@ static void ad1845_capture_format(struct snd_cs4231 * chip, struct snd_pcm_hw_pa
* try to support at least some of the extra bits by overriding
* some of the CS4231 callback.
*/
-static int __devinit create_ad1845(struct snd_card *card, unsigned port, int irq, int dma1)
+static int __devinit create_ad1845(struct snd_card *card, unsigned port,
+ int irq, int dma1, int dma2)
{
register struct soundscape *sscape = get_card_soundscape(card);
struct snd_cs4231 *chip;
int err;
-#define CS4231_SHARE_HARDWARE (CS4231_HWSHARE_DMA1 | CS4231_HWSHARE_DMA2)
- /*
- * The AD1845 PCM device is only half-duplex, and so
- * we only give it one DMA channel ...
- */
- if ((err = snd_cs4231_create(card,
- port, -1, irq, dma1, dma1,
- CS4231_HW_DETECT,
- CS4231_HWSHARE_DMA1, &chip)) == 0) {
+ if (sscape->type == SSCAPE_VIVO)
+ port += 4;
+
+ if (dma1 == dma2)
+ dma2 = -1;
+
+ err = snd_cs4231_create(card,
+ port, -1, irq, dma1, dma2,
+ CS4231_HW_DETECT, CS4231_HWSHARE_DMA1, &chip);
+ if (!err) {
unsigned long flags;
struct snd_pcm *pcm;
@@ -1031,49 +1071,72 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, int irq
snd_cs4231_mce_down(chip);
*/
- /*
- * The input clock frequency on the SoundScape must
- * be 14.31818 MHz, because we must set this register
- * to get the playback to sound correct ...
- */
- snd_cs4231_mce_up(chip);
- spin_lock_irqsave(&chip->reg_lock, flags);
- snd_cs4231_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20);
- spin_unlock_irqrestore(&chip->reg_lock, flags);
- snd_cs4231_mce_down(chip);
+ if (sscape->type != SSCAPE_VIVO) {
+ int val;
+ /*
+ * The input clock frequency on the SoundScape must
+ * be 14.31818 MHz, because we must set this register
+ * to get the playback to sound correct ...
+ */
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_cs4231_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_cs4231_mce_down(chip);
- /*
- * More custom configuration:
- * a) select "mode 2", and provide a current drive of 8 mA
- * b) enable frequency selection (for capture/playback)
- */
- spin_lock_irqsave(&chip->reg_lock, flags);
- snd_cs4231_out(chip, CS4231_MISC_INFO, (CS4231_MODE2 | 0x10));
- snd_cs4231_out(chip, AD1845_PWR_DOWN_CTRL, snd_cs4231_in(chip, AD1845_PWR_DOWN_CTRL) | AD1845_FREQ_SEL_ENABLE);
- spin_unlock_irqrestore(&chip->reg_lock, flags);
+ /*
+ * More custom configuration:
+ * a) select "mode 2" and provide a current drive of 8mA
+ * b) enable frequency selection (for capture/playback)
+ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_cs4231_out(chip, CS4231_MISC_INFO,
+ CS4231_MODE2 | 0x10);
+ val = snd_cs4231_in(chip, AD1845_PWR_DOWN_CTRL);
+ snd_cs4231_out(chip, AD1845_PWR_DOWN_CTRL,
+ val | AD1845_FREQ_SEL_ENABLE);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
- if ((err = snd_cs4231_pcm(chip, 0, &pcm)) < 0) {
- snd_printk(KERN_ERR "sscape: No PCM device for AD1845 chip\n");
+ err = snd_cs4231_pcm(chip, 0, &pcm);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: No PCM device "
+ "for AD1845 chip\n");
goto _error;
}
- if ((err = snd_cs4231_mixer(chip)) < 0) {
- snd_printk(KERN_ERR "sscape: No mixer device for AD1845 chip\n");
+ err = snd_cs4231_mixer(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: No mixer device "
+ "for AD1845 chip\n");
goto _error;
}
-
- if ((err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip))) < 0) {
- snd_printk(KERN_ERR "sscape: Could not create MIDI mixer control\n");
+ err = snd_cs4231_timer(chip, 0, NULL);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: No timer device "
+ "for AD1845 chip\n");
goto _error;
}
+ if (sscape->type != SSCAPE_VIVO) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&midi_mixer_ctl, chip));
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: Could not create "
+ "MIDI mixer control\n");
+ goto _error;
+ }
+ chip->set_playback_format = ad1845_playback_format;
+ chip->set_capture_format = ad1845_capture_format;
+ }
+
strcpy(card->driver, "SoundScape");
strcpy(card->shortname, pcm->name);
snprintf(card->longname, sizeof(card->longname),
- "%s at 0x%lx, IRQ %d, DMA %d\n",
- pcm->name, chip->port, chip->irq, chip->dma1);
- chip->set_playback_format = ad1845_playback_format;
- chip->set_capture_format = ad1845_capture_format;
+ "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n",
+ pcm->name, chip->port, chip->irq,
+ chip->dma1, chip->dma2);
+
sscape->chip = chip;
}
@@ -1086,15 +1149,15 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, int irq
* Create an ALSA soundcard entry for the SoundScape, using
* the given list of port, IRQ and DMA resources.
*/
-static int __devinit create_sscape(int dev, struct snd_card **rcardp)
+static int __devinit create_sscape(int dev, struct snd_card *card)
{
- struct snd_card *card;
- register struct soundscape *sscape;
- register unsigned dma_cfg;
+ struct soundscape *sscape = get_card_soundscape(card);
+ unsigned dma_cfg;
unsigned irq_cfg;
unsigned mpu_irq_cfg;
unsigned xport;
struct resource *io_res;
+ struct resource *wss_res;
unsigned long flags;
int err;
@@ -1118,61 +1181,69 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp)
* Grab IO ports that we will need to probe so that we
* can detect and control this hardware ...
*/
- if ((io_res = request_region(xport, 8, "SoundScape")) == NULL) {
+ io_res = request_region(xport, 8, "SoundScape");
+ if (!io_res) {
snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport);
return -EBUSY;
}
+ wss_res = NULL;
+ if (sscape->type == SSCAPE_VIVO) {
+ wss_res = request_region(wss_port[dev], 4, "SoundScape");
+ if (!wss_res) {
+ snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n",
+ wss_port[dev]);
+ err = -EBUSY;
+ goto _release_region;
+ }
+ }
/*
- * Grab both DMA channels (OK, only one for now) ...
+ * Grab one DMA channel ...
*/
- if ((err = request_dma(dma[dev], "SoundScape")) < 0) {
+ err = request_dma(dma[dev], "SoundScape");
+ if (err < 0) {
snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]);
goto _release_region;
}
- /*
- * Create a new ALSA sound card entry, in anticipation
- * of detecting our hardware ...
- */
- if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
- sizeof(struct soundscape))) == NULL) {
- err = -ENOMEM;
- goto _release_dma;
- }
-
- sscape = get_card_soundscape(card);
spin_lock_init(&sscape->lock);
spin_lock_init(&sscape->fwlock);
sscape->io_res = io_res;
+ sscape->wss_res = wss_res;
sscape->io_base = xport;
+ sscape->wss_base = wss_port[dev];
if (!detect_sscape(sscape)) {
printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base);
err = -ENODEV;
- goto _release_card;
+ goto _release_dma;
}
printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n",
- sscape->io_base, irq[dev], dma[dev]);
+ sscape->io_base, irq[dev], dma[dev]);
- /*
- * Now create the hardware-specific device so that we can
- * load the microcode into the on-board processor.
- * We cannot use the MPU-401 MIDI system until this firmware
- * has been loaded into the card.
- */
- if ((err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw))) < 0) {
- printk(KERN_ERR "sscape: Failed to create firmware device\n");
- goto _release_card;
+ if (sscape->type != SSCAPE_VIVO) {
+ /*
+ * Now create the hardware-specific device so that we can
+ * load the microcode into the on-board processor.
+ * We cannot use the MPU-401 MIDI system until this firmware
+ * has been loaded into the card.
+ */
+ err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw));
+ if (err < 0) {
+ printk(KERN_ERR "sscape: Failed to create "
+ "firmware device\n");
+ goto _release_dma;
+ }
+ strlcpy(sscape->hw->name, "SoundScape M68K",
+ sizeof(sscape->hw->name));
+ sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0';
+ sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE;
+ sscape->hw->ops.open = sscape_hw_open;
+ sscape->hw->ops.release = sscape_hw_release;
+ sscape->hw->ops.ioctl = sscape_hw_ioctl;
+ sscape->hw->private_data = sscape;
}
- strlcpy(sscape->hw->name, "SoundScape M68K", sizeof(sscape->hw->name));
- sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0';
- sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE;
- sscape->hw->ops.open = sscape_hw_open;
- sscape->hw->ops.release = sscape_hw_release;
- sscape->hw->ops.ioctl = sscape_hw_ioctl;
- sscape->hw->private_data = sscape;
/*
* Tell the on-board devices where their resources are (I think -
@@ -1197,7 +1268,8 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp)
sscape_write_unsafe(sscape->io_base,
GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg);
sscape_write_unsafe(sscape->io_base,
- GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1));
+ GA_CDCFG_REG, 0x09 | DMA_8BIT
+ | (dma[dev] << 4) | (irq_cfg << 1));
spin_unlock_irqrestore(&sscape->lock, flags);
@@ -1205,30 +1277,37 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp)
* We have now enabled the codec chip, and so we should
* detect the AD1845 device ...
*/
- if ((err = create_ad1845(card, CODEC_IO(xport), irq[dev], dma[dev])) < 0) {
- printk(KERN_ERR "sscape: No AD1845 device at 0x%x, IRQ %d\n",
- CODEC_IO(xport), irq[dev]);
- goto _release_card;
+ err = create_ad1845(card, wss_port[dev], irq[dev],
+ dma[dev], dma2[dev]);
+ if (err < 0) {
+ printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n",
+ wss_port[dev], irq[dev]);
+ goto _release_dma;
}
#define MIDI_DEVNUM 0
- if ((err = create_mpu401(card, MIDI_DEVNUM, MPU401_IO(xport), mpu_irq[dev])) < 0) {
- printk(KERN_ERR "sscape: Failed to create MPU-401 device at 0x%x\n",
- MPU401_IO(xport));
- goto _release_card;
- }
+ if (sscape->type != SSCAPE_VIVO) {
+ err = create_mpu401(card, MIDI_DEVNUM,
+ MPU401_IO(xport), mpu_irq[dev]);
+ if (err < 0) {
+ printk(KERN_ERR "sscape: Failed to create "
+ "MPU-401 device at 0x%x\n",
+ MPU401_IO(xport));
+ goto _release_dma;
+ }
- /*
- * Enable the master IRQ ...
- */
- sscape_write(sscape, GA_INTENA_REG, 0x80);
+ /*
+ * Enable the master IRQ ...
+ */
+ sscape_write(sscape, GA_INTENA_REG, 0x80);
- /*
- * Initialize mixer
- */
- sscape->midi_vol = 0;
- host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100);
- host_write_ctrl_unsafe(sscape->io_base, 0, 100);
- host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100);
+ /*
+ * Initialize mixer
+ */
+ sscape->midi_vol = 0;
+ host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100);
+ host_write_ctrl_unsafe(sscape->io_base, 0, 100);
+ host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100);
+ }
/*
* Now that we have successfully created this sound card,
@@ -1237,17 +1316,14 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp)
* function now that our "constructor" has completed.
*/
card->private_free = soundscape_free;
- *rcardp = card;
return 0;
- _release_card:
- snd_card_free(card);
-
- _release_dma:
+_release_dma:
free_dma(dma[dev]);
- _release_region:
+_release_region:
+ release_and_free_resource(wss_res);
release_and_free_resource(io_res);
return err;
@@ -1276,19 +1352,33 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i)
static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev)
{
struct snd_card *card;
+ struct soundscape *sscape;
int ret;
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct soundscape));
+ if (!card)
+ return -ENOMEM;
+
+ sscape = get_card_soundscape(card);
+ sscape->type = SSCAPE;
+
dma[dev] &= 0x03;
- ret = create_sscape(dev, &card);
+ ret = create_sscape(dev, card);
if (ret < 0)
- return ret;
+ goto _release_card;
+
snd_card_set_dev(card, pdev);
if ((ret = snd_card_register(card)) < 0) {
printk(KERN_ERR "sscape: Failed to register sound card\n");
- return ret;
+ goto _release_card;
}
dev_set_drvdata(pdev, card);
return 0;
+
+_release_card:
+ snd_card_free(card);
+ return ret;
}
static int __devexit snd_sscape_remove(struct device *devptr, unsigned int dev)
@@ -1325,6 +1415,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard,
static int idx = 0;
struct pnp_dev *dev;
struct snd_card *card;
+ struct soundscape *sscape;
int ret;
/*
@@ -1366,26 +1457,55 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard,
}
/*
+ * Create a new ALSA sound card entry, in anticipation
+ * of detecting our hardware ...
+ */
+ card = snd_card_new(index[idx], id[idx], THIS_MODULE,
+ sizeof(struct soundscape));
+ if (!card)
+ return -ENOMEM;
+
+ sscape = get_card_soundscape(card);
+
+ /*
+ * Identify card model ...
+ */
+ if (!strncmp("ENS4081", pid->id, 7))
+ sscape->type = SSCAPE_VIVO;
+ else
+ sscape->type = SSCAPE_PNP;
+
+ /*
* Read the correct parameters off the ISA PnP bus ...
*/
port[idx] = pnp_port_start(dev, 0);
irq[idx] = pnp_irq(dev, 0);
mpu_irq[idx] = pnp_irq(dev, 1);
dma[idx] = pnp_dma(dev, 0) & 0x03;
+ if (sscape->type == SSCAPE_PNP) {
+ dma2[idx] = dma[idx];
+ wss_port[idx] = CODEC_IO(port[idx]);
+ } else {
+ wss_port[idx] = pnp_port_start(dev, 1);
+ dma2[idx] = pnp_dma(dev, 1);
+ }
- ret = create_sscape(idx, &card);
+ ret = create_sscape(idx, card);
if (ret < 0)
- return ret;
+ goto _release_card;
+
snd_card_set_dev(card, &pcard->card->dev);
if ((ret = snd_card_register(card)) < 0) {
printk(KERN_ERR "sscape: Failed to register sound card\n");
- snd_card_free(card);
- return ret;
+ goto _release_card;
}
pnp_set_card_drvdata(pcard, card);
++idx;
+ return 0;
+_release_card:
+ snd_card_free(card);
return ret;
}
diff --git a/sound/isa/wavefront/Makefile b/sound/isa/wavefront/Makefile
index b4cb28422db0..601bdddd44d0 100644
--- a/sound/isa/wavefront/Makefile
+++ b/sound/isa/wavefront/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-wavefront-objs := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index bacc51c86587..a1ebb7c5c684 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -27,6 +27,7 @@
#include <linux/delay.h>
#include <linux/time.h>
#include <linux/wait.h>
+#include <linux/firmware.h>
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/snd_wavefront.h>
@@ -53,9 +54,8 @@ static int debug_default = 0; /* you can set this to control debugging
/* XXX this needs to be made firmware and hardware version dependent */
-static char *ospath = "/etc/sound/wavefront.os"; /* where to find a processed
- version of the WaveFront OS
- */
+#define DEFAULT_OSPATH "wavefront.os"
+static char *ospath = DEFAULT_OSPATH; /* the firmware file name */
static int wait_usecs = 150; /* This magic number seems to give pretty optimal
throughput based on my limited experimentation.
@@ -97,7 +97,7 @@ MODULE_PARM_DESC(sleep_interval, "how long to sleep when waiting for reply");
module_param(sleep_tries, int, 0444);
MODULE_PARM_DESC(sleep_tries, "how many times to try sleeping during a wait");
module_param(ospath, charp, 0444);
-MODULE_PARM_DESC(ospath, "full pathname to processed ICS2115 OS firmware");
+MODULE_PARM_DESC(ospath, "pathname to processed ICS2115 OS firmware");
module_param(reset_time, int, 0444);
MODULE_PARM_DESC(reset_time, "how long to wait for a reset to take effect");
module_param(ramcheck_time, int, 0444);
@@ -1768,7 +1768,7 @@ snd_wavefront_interrupt_bits (int irq)
static void __devinit
wavefront_should_cause_interrupt (snd_wavefront_t *dev,
- int val, int port, int timeout)
+ int val, int port, unsigned long timeout)
{
wait_queue_t wait;
@@ -1779,11 +1779,9 @@ wavefront_should_cause_interrupt (snd_wavefront_t *dev,
dev->irq_ok = 0;
outb (val,port);
spin_unlock_irq(&dev->irq_lock);
- while (1) {
- if ((timeout = schedule_timeout(timeout)) == 0)
- return;
- if (dev->irq_ok)
- return;
+ while (!dev->irq_ok && time_before(jiffies, timeout)) {
+ schedule_timeout_uninterruptible(1);
+ barrier();
}
}
@@ -1938,111 +1936,75 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev)
return (1);
}
-#include <linux/fs.h>
-#include <linux/mm.h>
-#include <linux/slab.h>
-#include <linux/unistd.h>
-#include <linux/syscalls.h>
-#include <asm/uaccess.h>
-
-
static int __devinit
wavefront_download_firmware (snd_wavefront_t *dev, char *path)
{
- unsigned char section[WF_SECTION_MAX];
- signed char section_length; /* yes, just a char; max value is WF_SECTION_MAX */
+ unsigned char *buf;
+ int len, err;
int section_cnt_downloaded = 0;
- int fd;
- int c;
- int i;
- mm_segment_t fs;
-
- /* This tries to be a bit cleverer than the stuff Alan Cox did for
- the generic sound firmware, in that it actually knows
- something about the structure of the Motorola firmware. In
- particular, it uses a version that has been stripped of the
- 20K of useless header information, and had section lengths
- added, making it possible to load the entire OS without any
- [kv]malloc() activity, since the longest entity we ever read is
- 42 bytes (well, WF_SECTION_MAX) long.
- */
-
- fs = get_fs();
- set_fs (get_ds());
+ const struct firmware *firmware;
- if ((fd = sys_open ((char __user *) path, 0, 0)) < 0) {
- snd_printk ("Unable to load \"%s\".\n",
- path);
+ err = request_firmware(&firmware, path, dev->card->dev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "firmware (%s) download failed!!!\n", path);
return 1;
}
- while (1) {
- int x;
-
- if ((x = sys_read (fd, (char __user *) &section_length, sizeof (section_length))) !=
- sizeof (section_length)) {
- snd_printk ("firmware read error.\n");
- goto failure;
- }
-
- if (section_length == 0) {
+ len = 0;
+ buf = firmware->data;
+ for (;;) {
+ int section_length = *(signed char *)buf;
+ if (section_length == 0)
break;
- }
-
if (section_length < 0 || section_length > WF_SECTION_MAX) {
- snd_printk ("invalid firmware section length %d\n",
- section_length);
+ snd_printk(KERN_ERR
+ "invalid firmware section length %d\n",
+ section_length);
goto failure;
}
+ buf++;
+ len++;
- if (sys_read (fd, (char __user *) section, section_length) != section_length) {
- snd_printk ("firmware section "
- "read error.\n");
+ if (firmware->size < len + section_length) {
+ snd_printk(KERN_ERR "firmware section read error.\n");
goto failure;
}
/* Send command */
-
- if (wavefront_write (dev, WFC_DOWNLOAD_OS)) {
+ if (wavefront_write(dev, WFC_DOWNLOAD_OS))
goto failure;
- }
- for (i = 0; i < section_length; i++) {
- if (wavefront_write (dev, section[i])) {
+ for (; section_length; section_length--) {
+ if (wavefront_write(dev, *buf))
goto failure;
- }
+ buf++;
+ len++;
}
/* get ACK */
-
- if (wavefront_wait (dev, STAT_CAN_READ)) {
-
- if ((c = inb (dev->data_port)) != WF_ACK) {
-
- snd_printk ("download "
- "of section #%d not "
- "acknowledged, ack = 0x%x\n",
- section_cnt_downloaded + 1, c);
- goto failure;
-
- }
-
- } else {
- snd_printk ("time out for firmware ACK.\n");
+ if (!wavefront_wait(dev, STAT_CAN_READ)) {
+ snd_printk(KERN_ERR "time out for firmware ACK.\n");
+ goto failure;
+ }
+ err = inb(dev->data_port);
+ if (err != WF_ACK) {
+ snd_printk(KERN_ERR
+ "download of section #%d not "
+ "acknowledged, ack = 0x%x\n",
+ section_cnt_downloaded + 1, err);
goto failure;
}
+ section_cnt_downloaded++;
}
- sys_close (fd);
- set_fs (fs);
+ release_firmware(firmware);
return 0;
failure:
- sys_close (fd);
- set_fs (fs);
- snd_printk ("firmware download failed!!!\n");
+ release_firmware(firmware);
+ snd_printk(KERN_ERR "firmware download failed!!!\n");
return 1;
}
@@ -2232,3 +2194,5 @@ snd_wavefront_detect (snd_wavefront_card_t *card)
return 0;
}
+
+MODULE_FIRMWARE(DEFAULT_OSPATH);
diff --git a/sound/last.c b/sound/last.c
index 964314efff5c..282b0cdb0589 100644
--- a/sound/last.c
+++ b/sound/last.c
@@ -1,6 +1,6 @@
/*
* Advanced Linux Sound Architecture
- * 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
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index c6b44102aa5b..356bf21a1506 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -170,14 +170,14 @@ config SND_CA0106
will be called snd-ca0106.
config SND_CMIPCI
- tristate "C-Media 8738, 8338"
+ tristate "C-Media 8338, 8738, 8768, 8770"
depends on SND
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_PCM
help
- If you want to use soundcards based on C-Media CMI8338 or CMI8738
- chips, say Y here and read
+ If you want to use soundcards based on C-Media CMI8338, CMI8738,
+ CMI8768 or CMI8770 chips, say Y here and read
<file:Documentation/sound/alsa/CMIPCI.txt>.
To compile this driver as a module, choose M here: the module
@@ -500,6 +500,103 @@ config SND_HDA_INTEL
To compile this driver as a module, choose M here: the module
will be called snd-hda-intel.
+config SND_HDA_HWDEP
+ bool "Build hwdep interface for HD-audio driver"
+ depends on SND_HDA_INTEL
+ select SND_HWDEP
+ help
+ Say Y here to build a hwdep interface for HD-audio driver.
+ This interface can be used for out-of-band communication
+ with codecs for debugging purposes.
+
+config SND_HDA_CODEC_REALTEK
+ bool "Build Realtek HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Realtek HD-audio codec support in
+ snd-hda-intel driver, such as ALC880.
+
+config SND_HDA_CODEC_ANALOG
+ bool "Build Analog Device HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Analog Device HD-audio codec support in
+ snd-hda-intel driver, such as AD1986A.
+
+config SND_HDA_CODEC_SIGMATEL
+ bool "Build IDT/Sigmatel HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include IDT (Sigmatel) HD-audio codec support in
+ snd-hda-intel driver, such as STAC9200.
+
+config SND_HDA_CODEC_VIA
+ bool "Build VIA HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include VIA HD-audio codec support in
+ snd-hda-intel driver, such as VT1708.
+
+config SND_HDA_CODEC_ATIHDMI
+ bool "Build ATI HDMI HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include ATI HDMI HD-audio codec support in
+ snd-hda-intel driver, such as ATI RS600 HDMI.
+
+config SND_HDA_CODEC_CONEXANT
+ bool "Build Conexant HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Conexant HD-audio codec support in
+ snd-hda-intel driver, such as CX20549.
+
+config SND_HDA_CODEC_CMEDIA
+ bool "Build C-Media HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include C-Media HD-audio codec support in
+ snd-hda-intel driver, such as CMI9880.
+
+config SND_HDA_CODEC_SI3054
+ bool "Build Silicon Labs 3054 HD-modem codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Silicon Labs 3054 HD-modem codec
+ (and compatibles) support in snd-hda-intel driver.
+
+config SND_HDA_GENERIC
+ bool "Enable generic HD-audio codec parser"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to enable the generic HD-audio codec parser
+ in snd-hda-intel driver.
+
+config SND_HDA_POWER_SAVE
+ bool "Aggressive power-saving on HD-audio"
+ depends on SND_HDA_INTEL && EXPERIMENTAL
+ help
+ Say Y here to enable more aggressive power-saving mode on
+ HD-audio driver. The power-saving timeout can be configured
+ via power_save option or over sysfs on-the-fly.
+
+config SND_HDA_POWER_SAVE_DEFAULT
+ int "Default time-out for HD-audio power-save mode"
+ depends on SND_HDA_POWER_SAVE
+ default 0
+ help
+ The default time-out value in seconds for HD-audio automatic
+ power-save mode. 0 means to disable the power-save mode.
+
config SND_HDSP
tristate "RME Hammerfall DSP Audio"
depends on SND
@@ -799,4 +896,12 @@ config SND_AC97_POWER_SAVE
snd-ac97-codec driver. You can toggle it dynamically over
sysfs, too.
+config SND_AC97_POWER_SAVE_DEFAULT
+ int "Default time-out for AC97 power-save mode"
+ depends on SND_AC97_POWER_SAVE
+ default 0
+ help
+ The default time-out value in seconds for AC97 automatic
+ power-save mode. 0 means to disable the power-save mode.
+
endmenu
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index cd76e0293d06..09ddc82eeca2 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ad1889-objs := ad1889.o
diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile
index f5d471896b95..0be48b1a22d0 100644
--- a/sound/pci/ac97/Makefile
+++ b/sound/pci/ac97/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index bbed644bf9c5..6a9966df0cc9 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1,5 +1,5 @@
/*
- * 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 2.2
@@ -39,7 +39,7 @@
#include "ac97_patch.c"
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Universal interface for Audio Codec '97");
MODULE_LICENSE("GPL");
@@ -49,7 +49,7 @@ module_param(enable_loopback, bool, 0444);
MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control");
#ifdef CONFIG_SND_AC97_POWER_SAVE
-static int power_save;
+static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT;
module_param(power_save, bool, 0644);
MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control");
#endif
@@ -176,7 +176,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL},
{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL},
{ 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF},
-{ 0x594d4800, 0xffffffff, "YMF743", NULL, NULL },
+{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL },
{ 0x594d4802, 0xffffffff, "YMF752", NULL, NULL },
{ 0x594d4803, 0xffffffff, "YMF753", patch_yamaha_ymf753, NULL },
{ 0x83847600, 0xffffffff, "STAC9700,83,84", patch_sigmatel_stac9700, NULL },
@@ -779,6 +779,12 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_
change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC,
AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT,
v);
+ } else if (ac97->id == AC97_ID_YMF743) {
+ change |= snd_ac97_update_bits_nolock(ac97,
+ AC97_YMF7X3_DIT_CTRL,
+ 0xff38,
+ ((val << 4) & 0xff00) |
+ ((val << 2) & 0x0038));
} else {
unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS);
snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */
@@ -1375,7 +1381,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
for (idx = 0; idx < 2; idx++) {
if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0)
return err;
- if (ac97->id == AC97_ID_YMF753) {
+ if (ac97->id == AC97_ID_YMF743 ||
+ ac97->id == AC97_ID_YMF753) {
kctl->private_value &= ~(0xff << 16);
kctl->private_value |= 7 << 16;
}
@@ -2036,11 +2043,12 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
else {
udelay(50);
if (ac97->scaps & AC97_SCAP_SKIP_AUDIO)
- err = ac97_reset_wait(ac97, HZ/2, 1);
+ err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 1);
else {
- err = ac97_reset_wait(ac97, HZ/2, 0);
+ err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 0);
if (err < 0)
- err = ac97_reset_wait(ac97, HZ/2, 1);
+ err = ac97_reset_wait(ac97,
+ msecs_to_jiffies(500), 1);
}
if (err < 0) {
snd_printk(KERN_WARNING "AC'97 %d does not respond - RESET\n", ac97->num);
@@ -2104,7 +2112,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
}
/* nothing should be in powerdown mode */
snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0);
- end_time = jiffies + (HZ / 10);
+ end_time = jiffies + msecs_to_jiffies(100);
do {
if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
goto __ready_ok;
@@ -2136,7 +2144,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
udelay(100);
/* nothing should be in powerdown mode */
snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0);
- end_time = jiffies + (HZ / 10);
+ end_time = jiffies + msecs_to_jiffies(100);
do {
if ((snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS) & tmp) == tmp)
goto __ready_ok;
@@ -2354,7 +2362,8 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup)
* (for avoiding loud click noises for many (OSS) apps
* that open/close frequently)
*/
- schedule_delayed_work(&ac97->power_work, HZ*2);
+ schedule_delayed_work(&ac97->power_work,
+ msecs_to_jiffies(2000));
else {
cancel_delayed_work(&ac97->power_work);
update_power_regs(ac97);
@@ -2436,7 +2445,7 @@ EXPORT_SYMBOL(snd_ac97_suspend);
/*
* restore ac97 status
*/
-void snd_ac97_restore_status(struct snd_ac97 *ac97)
+static void snd_ac97_restore_status(struct snd_ac97 *ac97)
{
int i;
@@ -2457,7 +2466,7 @@ void snd_ac97_restore_status(struct snd_ac97 *ac97)
/*
* restore IEC958 status
*/
-void snd_ac97_restore_iec958(struct snd_ac97 *ac97)
+static void snd_ac97_restore_iec958(struct snd_ac97 *ac97)
{
if (ac97->ext_id & AC97_EI_SPDIF) {
if (ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_SPDIF) {
@@ -2494,7 +2503,10 @@ void snd_ac97_resume(struct snd_ac97 *ac97)
snd_ac97_write(ac97, AC97_POWERDOWN, 0);
if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) {
- snd_ac97_write(ac97, AC97_RESET, 0);
+ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO))
+ snd_ac97_write(ac97, AC97_RESET, 0);
+ else if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM))
+ snd_ac97_write(ac97, AC97_EXTENDED_MID, 0);
udelay(100);
snd_ac97_write(ac97, AC97_POWERDOWN, 0);
}
diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h
index 6d73514dc49e..c129492c82b3 100644
--- a/sound/pci/ac97/ac97_id.h
+++ b/sound/pci/ac97/ac97_id.h
@@ -1,5 +1,5 @@
/*
- * 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 2.2
@@ -54,6 +54,7 @@
#define AC97_ID_ALC658 0x414c4780
#define AC97_ID_ALC658D 0x414c4781
#define AC97_ID_ALC850 0x414c4790
+#define AC97_ID_YMF743 0x594d4800
#define AC97_ID_YMF753 0x594d4803
#define AC97_ID_VT1616 0x49434551
#define AC97_ID_CM9738 0x434d4941
diff --git a/sound/pci/ac97/ac97_local.h b/sound/pci/ac97/ac97_local.h
index 78745c5c6df8..c276a5e3f7ac 100644
--- a/sound/pci/ac97/ac97_local.h
+++ b/sound/pci/ac97/ac97_local.h
@@ -1,5 +1,5 @@
/*
- * 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 2.2
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 581ebba4d1a7..98c8b727b62b 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1,5 +1,5 @@
/*
- * 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 2.2
@@ -204,9 +204,13 @@ static inline int is_shared_micin(struct snd_ac97 *ac97)
/* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */
+/* Modified for YMF743 by Keita Maehara <maehara@debian.org> */
-/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */
-static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+/* It is possible to indicate to the Yamaha YMF7x3 the type of
+ speakers being used. */
+
+static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
static char *texts[3] = {
"Standard", "Small", "Smaller"
@@ -221,12 +225,13 @@ static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct sn
return 0;
}
-static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = ac97->regs[AC97_YMF753_3D_MODE_SEL];
+ val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL];
val = (val >> 10) & 3;
if (val > 0) /* 0 = invalid */
val--;
@@ -234,7 +239,8 @@ static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd
return 0;
}
-static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
@@ -242,20 +248,22 @@ static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd
if (ucontrol->value.enumerated.item[0] > 2)
return -EINVAL;
val = (ucontrol->value.enumerated.item[0] + 1) << 10;
- return snd_ac97_update(ac97, AC97_YMF753_3D_MODE_SEL, val);
+ return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val);
}
-static const struct snd_kcontrol_new snd_ac97_ymf753_controls_speaker =
+static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "3D Control - Speaker",
- .info = snd_ac97_ymf753_info_speaker,
- .get = snd_ac97_ymf753_get_speaker,
- .put = snd_ac97_ymf753_put_speaker,
+ .info = snd_ac97_ymf7x3_info_speaker,
+ .get = snd_ac97_ymf7x3_get_speaker,
+ .put = snd_ac97_ymf7x3_put_speaker,
};
-/* It is possible to indicate to the Yamaha YMF753 the source to direct to the S/PDIF output. */
-static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+/* It is possible to indicate to the Yamaha YMF7x3 the source to
+ direct to the S/PDIF output. */
+static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
static char *texts[2] = { "AC-Link", "A/D Converter" };
@@ -268,17 +276,19 @@ static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, stru
return 0;
}
-static int snd_ac97_ymf753_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = ac97->regs[AC97_YMF753_DIT_CTRL2];
+ val = ac97->regs[AC97_YMF7X3_DIT_CTRL];
ucontrol->value.enumerated.item[0] = (val >> 1) & 1;
return 0;
}
-static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
@@ -286,7 +296,75 @@ static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struc
if (ucontrol->value.enumerated.item[0] > 1)
return -EINVAL;
val = ucontrol->value.enumerated.item[0] << 1;
- return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0002, val);
+ return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val);
+}
+
+static int patch_yamaha_ymf7x3_3d(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97);
+ err = snd_ctl_add(ac97->bus->card, kctl);
+ if (err < 0)
+ return err;
+ strcpy(kctl->id.name, "3D Control - Wide");
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0);
+ snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
+ err = snd_ctl_add(ac97->bus->card,
+ snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker,
+ ac97));
+ if (err < 0)
+ return err;
+ snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00);
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_yamaha_ymf743_controls_spdif[3] =
+{
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+ AC97_YMF7X3_DIT_CTRL, 0, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Source",
+ .info = snd_ac97_ymf7x3_spdif_source_info,
+ .get = snd_ac97_ymf7x3_spdif_source_get,
+ .put = snd_ac97_ymf7x3_spdif_source_put,
+ },
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute",
+ AC97_YMF7X3_DIT_CTRL, 2, 1, 1)
+};
+
+static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97)
+{
+ int err;
+
+ err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3);
+ if (err < 0)
+ return err;
+ err = patch_build_controls(ac97,
+ snd_ac97_yamaha_ymf743_controls_spdif, 3);
+ if (err < 0)
+ return err;
+ /* set default PCM S/PDIF params */
+ /* PCM audio,no copyright,no preemphasis,PCM coder,original */
+ snd_ac97_write_cache(ac97, AC97_YMF7X3_DIT_CTRL, 0xa201);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = {
+ .build_spdif = patch_yamaha_ymf743_build_spdif,
+ .build_3d = patch_yamaha_ymf7x3_3d,
+};
+
+static int patch_yamaha_ymf743(struct snd_ac97 *ac97)
+{
+ ac97->build_ops = &patch_yamaha_ymf743_ops;
+ ac97->caps |= AC97_BC_BASS_TREBLE;
+ ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
+ return 0;
}
/* The AC'97 spec states that the S/PDIF signal is to be output at pin 48.
@@ -311,7 +389,7 @@ static int snd_ac97_ymf753_spdif_output_pin_get(struct snd_kcontrol *kcontrol, s
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = ac97->regs[AC97_YMF753_DIT_CTRL2];
+ val = ac97->regs[AC97_YMF7X3_DIT_CTRL];
ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0;
return 0;
}
@@ -325,7 +403,7 @@ static int snd_ac97_ymf753_spdif_output_pin_put(struct snd_kcontrol *kcontrol, s
return -EINVAL;
val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 :
(ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0;
- return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0028, val);
+ return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val);
/* The following can be used to direct S/PDIF output to pin 47 (EAPD).
snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */
}
@@ -334,9 +412,9 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = snd_ac97_ymf753_spdif_source_info,
- .get = snd_ac97_ymf753_spdif_source_get,
- .put = snd_ac97_ymf753_spdif_source_put,
+ .info = snd_ac97_ymf7x3_spdif_source_info,
+ .get = snd_ac97_ymf7x3_spdif_source_get,
+ .put = snd_ac97_ymf7x3_spdif_source_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -345,25 +423,10 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = {
.get = snd_ac97_ymf753_spdif_output_pin_get,
.put = snd_ac97_ymf753_spdif_output_pin_put,
},
- AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF753_DIT_CTRL2, 2, 1, 1)
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute",
+ AC97_YMF7X3_DIT_CTRL, 2, 1, 1)
};
-static int patch_yamaha_ymf753_3d(struct snd_ac97 * ac97)
-{
- struct snd_kcontrol *kctl;
- int err;
-
- if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0)
- return err;
- strcpy(kctl->id.name, "3D Control - Wide");
- kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0);
- snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
- if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf753_controls_speaker, ac97))) < 0)
- return err;
- snd_ac97_write_cache(ac97, AC97_YMF753_3D_MODE_SEL, 0x0c00);
- return 0;
-}
-
static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97)
{
int err;
@@ -374,7 +437,7 @@ static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97)
}
static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = {
- .build_3d = patch_yamaha_ymf753_3d,
+ .build_3d = patch_yamaha_ymf7x3_3d,
.build_post_spdif = patch_yamaha_ymf753_post_spdif
};
@@ -1880,14 +1943,7 @@ static int patch_ad1981b(struct snd_ac97 *ac97)
return 0;
}
-static int snd_ac97_ad1888_lohpsel_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 snd_ac97_ad1888_lohpsel_info snd_ctl_boolean_mono_info
static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2186,15 +2242,7 @@ static int patch_ad1985(struct snd_ac97 * ac97)
return 0;
}
-static int snd_ac97_ad1986_bool_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 snd_ac97_ad1986_bool_info snd_ctl_boolean_mono_info
static int snd_ac97_ad1986_lososel_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h
index fd341ce63762..9cccc27ea1b5 100644
--- a/sound/pci/ac97/ac97_patch.h
+++ b/sound/pci/ac97/ac97_patch.h
@@ -1,5 +1,5 @@
/*
- * 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 2.2
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
index 4281e6d0c5b6..8cbc03332b01 100644
--- a/sound/pci/ac97/ac97_pcm.c
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -1,5 +1,5 @@
/*
- * 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 2.2
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
index a3fdd7da911c..fed4a2c3d8a1 100644
--- a/sound/pci/ac97/ac97_proc.c
+++ b/sound/pci/ac97/ac97_proc.c
@@ -1,5 +1,5 @@
/*
- * 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 2.2
@@ -236,10 +236,14 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe
val = snd_ac97_read(ac97, AC97_PCM_MIC_ADC_RATE);
snd_iprintf(buffer, "PCM MIC ADC : %iHz\n", val);
}
- if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF)) {
+ if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF) ||
+ (ac97->id == AC97_ID_YMF743)) {
if (ac97->flags & AC97_CS_SPDIF)
val = snd_ac97_read(ac97, AC97_CSR_SPDIF);
- else
+ else if (ac97->id == AC97_ID_YMF743) {
+ val = snd_ac97_read(ac97, AC97_YMF7X3_DIT_CTRL);
+ val = 0x2000 | (val & 0xff00) >> 4 | (val & 0x38) >> 2;
+ } else
val = snd_ac97_read(ac97, AC97_SPDIF);
snd_iprintf(buffer, "SPDIF Control :%s%s%s%s Category=0x%x Generation=%i%s%s%s\n",
diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c
index dc26820a03a5..722de451d15f 100644
--- a/sound/pci/ac97/ak4531_codec.c
+++ b/sound/pci/ac97/ak4531_codec.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal routines for AK4531 codec
*
*
@@ -29,7 +29,7 @@
#include <sound/ak4531_codec.h>
#include <sound/tlv.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Universal routines for AK4531 codec");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/ali5451/Makefile b/sound/pci/ali5451/Makefile
index 2e1831597474..713459c12d22 100644
--- a/sound/pci/ali5451/Makefile
+++ b/sound/pci/ali5451/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ali5451-objs := ali5451.o
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 05b4c8696941..4c2bd7adf674 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1804,15 +1804,7 @@ static int __devinit snd_ali_build_pcms(struct snd_ali *codec)
.info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \
.put = snd_ali5451_spdif_put, .private_value = value}
-static int snd_ali5451_spdif_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 snd_ali5451_spdif_info snd_ctl_boolean_mono_info
static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 8fb55d3b454b..1190ef366a41 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -1,7 +1,7 @@
/*
* card-als4000.c - driver for Avance Logic ALS4000 based soundcards.
* Copyright (C) 2000 by Bart Hartgers <bart@etpmod.phys.tue.nl>,
- * Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
* Copyright (C) 2002 by Andreas Mohr <hw7oshyuv3001@sneakemail.com>
*
* Framework borrowed from Massimo Piccioni's card-als100.c.
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 5ec1b6fcd548..f70286a7364a 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -232,6 +232,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
pci_disable_device(chip->pci_dev);
//FIXME: this not the right place to unregister the gameport
vortex_gameport_unregister(chip);
+ kfree(chip);
return err;
}
diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c
index 0c86a31c4336..38602b85874d 100644
--- a/sound/pci/au88x0/au88x0_eq.c
+++ b/sound/pci/au88x0/au88x0_eq.c
@@ -728,15 +728,7 @@ static void vortex_Eqlzr_shutdown(vortex_t * vortex)
/* ALSA interface */
/* Control interface */
-static int
-snd_vortex_eqtoggle_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 snd_vortex_eqtoggle_info snd_ctl_boolean_mono_info
static int
snd_vortex_eqtoggle_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
index c75d368ea087..8db3d3e6f7bb 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of MPU-401 in UART mode
*
* Modified for the Aureal Vortex based Soundcards
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index d3e662a1285d..978b856f5621 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -370,8 +370,8 @@ static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr)
while ((edx & 0x80000000) == 0) {
edx <<= 1;
eax--;
- if (eax == 0) ;
- break;
+ if (eax == 0)
+ break;
}
if (eax)
edx <<= 1;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 131952f55857..91f9e6a112ff 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -147,15 +147,56 @@ MODULE_PARM_DESC(load_all, "Allow to load the non-whitelisted cards");
/* SYNC, one WRITE per line, one extra WRITE per page boundary, SYNC, JUMP */
#define MAX_RISC_SIZE ((1 + 255 + (PAGE_ALIGN(255 * 4092) / PAGE_SIZE - 1) + 1 + 1) * 8)
+/* Cards with configuration information */
+enum snd_bt87x_boardid {
+ SND_BT87X_BOARD_UNKNOWN,
+ SND_BT87X_BOARD_GENERIC, /* both an & dig interfaces, 32kHz */
+ SND_BT87X_BOARD_ANALOG, /* board with no external A/D */
+ SND_BT87X_BOARD_OSPREY2x0,
+ SND_BT87X_BOARD_OSPREY440,
+ SND_BT87X_BOARD_AVPHONE98,
+};
+
+/* Card configuration */
+struct snd_bt87x_board {
+ int dig_rate; /* Digital input sampling rate */
+ u32 digital_fmt; /* Register settings for digital input */
+ unsigned no_analog:1; /* No analog input */
+ unsigned no_digital:1; /* No digital input */
+};
+
+static const __devinitdata struct snd_bt87x_board snd_bt87x_boards[] = {
+ [SND_BT87X_BOARD_UNKNOWN] = {
+ .dig_rate = 32000, /* just a guess */
+ },
+ [SND_BT87X_BOARD_GENERIC] = {
+ .dig_rate = 32000,
+ },
+ [SND_BT87X_BOARD_ANALOG] = {
+ .no_digital = 1,
+ },
+ [SND_BT87X_BOARD_OSPREY2x0] = {
+ .dig_rate = 44100,
+ .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT),
+ },
+ [SND_BT87X_BOARD_OSPREY440] = {
+ .dig_rate = 32000,
+ .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT),
+ .no_analog = 1,
+ },
+ [SND_BT87X_BOARD_AVPHONE98] = {
+ .dig_rate = 48000,
+ },
+};
+
struct snd_bt87x {
struct snd_card *card;
struct pci_dev *pci;
+ struct snd_bt87x_board board;
void __iomem *mmio;
int irq;
- int dig_rate;
-
spinlock_t reg_lock;
unsigned long opened;
struct snd_pcm_substream *substream;
@@ -340,30 +381,11 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
static int snd_bt87x_set_digital_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime)
{
- static struct {
- int rate;
- unsigned int bit;
- } ratebits[] = {
- {8000, SNDRV_PCM_RATE_8000},
- {11025, SNDRV_PCM_RATE_11025},
- {16000, SNDRV_PCM_RATE_16000},
- {22050, SNDRV_PCM_RATE_22050},
- {32000, SNDRV_PCM_RATE_32000},
- {44100, SNDRV_PCM_RATE_44100},
- {48000, SNDRV_PCM_RATE_48000}
- };
- int i;
-
- chip->reg_control |= CTL_DA_IOM_DA;
+ chip->reg_control |= CTL_DA_IOM_DA | CTL_A_PWRDN;
runtime->hw = snd_bt87x_digital_hw;
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
- for (i = 0; i < ARRAY_SIZE(ratebits); ++i)
- if (chip->dig_rate == ratebits[i].rate) {
- runtime->hw.rates = ratebits[i].bit;
- break;
- }
- runtime->hw.rate_min = chip->dig_rate;
- runtime->hw.rate_max = chip->dig_rate;
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->board.dig_rate);
+ runtime->hw.rate_min = chip->board.dig_rate;
+ runtime->hw.rate_max = chip->board.dig_rate;
return 0;
}
@@ -380,7 +402,7 @@ static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_runtim
.rats = &analog_clock
};
- chip->reg_control &= ~CTL_DA_IOM_DA;
+ chip->reg_control &= ~(CTL_DA_IOM_DA | CTL_A_PWRDN);
runtime->hw = snd_bt87x_analog_hw;
return snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&constraint_rates);
@@ -419,6 +441,11 @@ static int snd_bt87x_close(struct snd_pcm_substream *substream)
{
struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+ spin_lock_irq(&chip->reg_lock);
+ chip->reg_control |= CTL_A_PWRDN;
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ spin_unlock_irq(&chip->reg_lock);
+
chip->substream = NULL;
clear_bit(0, &chip->opened);
smp_mb__after_clear_bit();
@@ -569,15 +596,7 @@ static struct snd_kcontrol_new snd_bt87x_capture_volume = {
.put = snd_bt87x_capture_volume_put,
};
-static int snd_bt87x_capture_boost_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *info)
-{
- info->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- info->count = 1;
- info->value.integer.min = 0;
- info->value.integer.max = 1;
- return 0;
-}
+#define snd_bt87x_capture_boost_info snd_ctl_boolean_mono_info
static int snd_bt87x_capture_boost_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *value)
@@ -736,61 +755,69 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
pci_resource_len(pci, 0));
if (!chip->mmio) {
- snd_bt87x_free(chip);
snd_printk(KERN_ERR "cannot remap io memory\n");
- return -ENOMEM;
+ err = -ENOMEM;
+ goto fail;
}
- chip->reg_control = CTL_DA_ES2 | CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT);
+ chip->reg_control = CTL_A_PWRDN | CTL_DA_ES2 |
+ CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT);
chip->interrupt_mask = MY_INTERRUPTS;
snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
snd_bt87x_writel(chip, REG_INT_MASK, 0);
snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
- if (request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
- "Bt87x audio", chip)) {
- snd_bt87x_free(chip);
- snd_printk(KERN_ERR "cannot grab irq\n");
- return -EBUSY;
+ err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
+ "Bt87x audio", chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
+ goto fail;
}
chip->irq = pci->irq;
pci_set_master(pci);
synchronize_irq(chip->irq);
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
- if (err < 0) {
- snd_bt87x_free(chip);
- return err;
- }
+ if (err < 0)
+ goto fail;
+
snd_card_set_dev(card, &pci->dev);
*rchip = chip;
return 0;
+
+fail:
+ snd_bt87x_free(chip);
+ return err;
}
-#define BT_DEVICE(chip, subvend, subdev, rate) \
+#define BT_DEVICE(chip, subvend, subdev, id) \
{ .vendor = PCI_VENDOR_ID_BROOKTREE, \
.device = chip, \
.subvendor = subvend, .subdevice = subdev, \
- .driver_data = rate }
+ .driver_data = SND_BT87X_BOARD_ ## id }
+/* driver_data is the card id for that device */
-/* driver_data is the default digital_rate value for that device */
static struct pci_device_id snd_bt87x_ids[] = {
/* Hauppauge WinTV series */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC),
/* Hauppauge WinTV series */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, GENERIC),
/* Viewcast Osprey 200 */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, OSPREY2x0),
/* Viewcast Osprey 440 (rate is configurable via gpio) */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, OSPREY440),
/* ATI TV-Wonder */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
/* Leadtek Winfast tv 2000xp delux */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
/* Voodoo TV 200 */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
/* AVerMedia Studio No. 103, 203, ...? */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, AVPHONE98),
+ /* Prolink PixelView PV-M4900 */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1554, 0x4011, GENERIC),
+ /* Pinnacle Studio PCTV rave */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0xbd11, 0x1200, GENERIC),
{ }
};
MODULE_DEVICE_TABLE(pci, snd_bt87x_ids);
@@ -815,7 +842,7 @@ static struct {
static struct pci_driver driver;
-/* return the rate of the card, or a negative value if it's blacklisted */
+/* return the id of the card, or a negative value if it's blacklisted */
static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
{
int i;
@@ -833,12 +860,12 @@ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
return -EBUSY;
}
- snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x, using default rate 32000\n",
- pci->device, pci->subsystem_vendor, pci->subsystem_device);
+ snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x\n",
+ pci->device, pci->subsystem_vendor, pci->subsystem_device);
snd_printk(KERN_DEBUG "please mail id, board name, and, "
"if it works, the correct digital_rate option to "
"<alsa-devel@alsa-project.org>\n");
- return 32000; /* default rate */
+ return SND_BT87X_BOARD_UNKNOWN;
}
static int __devinit snd_bt87x_probe(struct pci_dev *pci,
@@ -847,12 +874,16 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci,
static int dev;
struct snd_card *card;
struct snd_bt87x *chip;
- int err, rate;
+ int err;
+ enum snd_bt87x_boardid boardid;
- rate = pci_id->driver_data;
- if (! rate)
- if ((rate = snd_bt87x_detect_card(pci)) <= 0)
+ if (!pci_id->driver_data) {
+ err = snd_bt87x_detect_card(pci);
+ if (err < 0)
return -ENODEV;
+ boardid = err;
+ } else
+ boardid = pci_id->driver_data;
if (dev >= SNDRV_CARDS)
return -ENODEV;
@@ -869,27 +900,39 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci,
if (err < 0)
goto _error;
- if (digital_rate[dev] > 0)
- chip->dig_rate = digital_rate[dev];
- else
- chip->dig_rate = rate;
+ memcpy(&chip->board, &snd_bt87x_boards[boardid], sizeof(chip->board));
- err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital");
- if (err < 0)
- goto _error;
- err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog");
- if (err < 0)
- goto _error;
+ if (!chip->board.no_digital) {
+ if (digital_rate[dev] > 0)
+ chip->board.dig_rate = digital_rate[dev];
- err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_volume, chip));
- if (err < 0)
- goto _error;
- err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_boost, chip));
- if (err < 0)
- goto _error;
- err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_source, chip));
- if (err < 0)
- goto _error;
+ chip->reg_control |= chip->board.digital_fmt;
+
+ err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital");
+ if (err < 0)
+ goto _error;
+ }
+ if (!chip->board.no_analog) {
+ err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog");
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_volume, chip));
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_boost, chip));
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_source, chip));
+ if (err < 0)
+ goto _error;
+ }
+ snd_printk(KERN_INFO "bt87x%d: Using board %d, %sanalog, %sdigital "
+ "(rate %d Hz)\n", dev, boardid,
+ chip->board.no_analog ? "no " : "",
+ chip->board.no_digital ? "no " : "", chip->board.dig_rate);
strcpy(card->driver, "Bt87x");
sprintf(card->shortname, "Brooktree Bt%x", pci->device);
@@ -920,8 +963,8 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci)
/* default entries for all Bt87x cards - it's not exported */
/* driver_data is set to 0 to call detection */
static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = {
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, 0),
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, 0),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN),
{ }
};
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index a0420bc63f0b..75da1746e758 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.21
+ * Version: 0.0.22
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -47,6 +47,8 @@
* Added GPIO info for SB Live 24bit.
* 0.0.21
* Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.22
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
*
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
@@ -552,6 +554,95 @@
#define CONTROL_CENTER_LFE_CHANNEL 1
#define CONTROL_UNKNOWN_CHANNEL 2
+
+/* Based on WM8768 Datasheet Rev 4.2 page 32 */
+#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */
+#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */
+
+#define SPI_LDA1_REG 0 /* digital attenuation */
+#define SPI_RDA1_REG 1
+#define SPI_LDA2_REG 4
+#define SPI_RDA2_REG 5
+#define SPI_LDA3_REG 6
+#define SPI_RDA3_REG 7
+#define SPI_LDA4_REG 13
+#define SPI_RDA4_REG 14
+#define SPI_MASTDA_REG 8
+
+#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */
+#define SPI_DA_BIT_0dB 0xff /* 0 dB */
+#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */
+
+#define SPI_PL_REG 2
+#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */
+#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */
+#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */
+#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */
+#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */
+#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */
+#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */
+#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */
+#define SPI_IZD_REG 2
+#define SPI_IZD_BIT (1<<4) /* infinite zero detect */
+
+#define SPI_FMT_REG 3
+#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */
+#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */
+#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */
+#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */
+#define SPI_LRP_REG 3
+#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */
+#define SPI_BCP_REG 3
+#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */
+#define SPI_IWL_REG 3
+#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */
+#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */
+#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */
+#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */
+
+#define SPI_MS_REG 10
+#define SPI_MS_BIT (1<<5) /* master mode */
+#define SPI_RATE_REG 10 /* only applies in master mode */
+#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */
+#define SPI_RATE_BIT_192 (1<<6)
+#define SPI_RATE_BIT_256 (2<<6)
+#define SPI_RATE_BIT_384 (3<<6)
+#define SPI_RATE_BIT_512 (4<<6)
+#define SPI_RATE_BIT_768 (5<<6)
+
+/* They really do label the bit for the 4th channel "4" and not "3" */
+#define SPI_DMUTE0_REG 9
+#define SPI_DMUTE1_REG 9
+#define SPI_DMUTE2_REG 9
+#define SPI_DMUTE4_REG 15
+#define SPI_DMUTE0_BIT (1<<3)
+#define SPI_DMUTE1_BIT (1<<4)
+#define SPI_DMUTE2_BIT (1<<5)
+#define SPI_DMUTE4_BIT (1<<2)
+
+#define SPI_PHASE0_REG 3
+#define SPI_PHASE1_REG 3
+#define SPI_PHASE2_REG 3
+#define SPI_PHASE4_REG 15
+#define SPI_PHASE0_BIT (1<<6)
+#define SPI_PHASE1_BIT (1<<7)
+#define SPI_PHASE2_BIT (1<<8)
+#define SPI_PHASE4_BIT (1<<3)
+
+#define SPI_PDWN_REG 2 /* power down all DACs */
+#define SPI_PDWN_BIT (1<<2)
+#define SPI_DACD0_REG 10 /* power down individual DACs */
+#define SPI_DACD1_REG 10
+#define SPI_DACD2_REG 10
+#define SPI_DACD4_REG 15
+#define SPI_DACD0_BIT (1<<1)
+#define SPI_DACD1_BIT (1<<2)
+#define SPI_DACD2_BIT (1<<3)
+#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */
+
+#define SPI_PWRDNALL_REG 10 /* power down everything */
+#define SPI_PWRDNALL_BIT (1<<4)
+
#include "ca_midi.h"
struct snd_ca0106;
@@ -611,6 +702,8 @@ struct snd_ca0106 {
struct snd_ca_midi midi;
struct snd_ca_midi midi2;
+
+ u16 spi_dac_reg[16];
};
int snd_ca0106_mixer(struct snd_ca0106 *emu);
@@ -627,4 +720,5 @@ void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
-
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+ unsigned int data);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index fcab8fb97e38..31d8db9f7a4c 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.23
+ * Version: 0.0.25
*
* FEATURES currently supported:
* Front, Rear and Center/LFE.
@@ -79,6 +79,10 @@
* Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901
* 0.0.23
* Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.24
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ * 0.0.25
+ * Powerdown SPI DAC channels when not in use
*
* BUGS:
* Some stability problems when unloading the snd-ca0106 kernel module.
@@ -170,6 +174,15 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
static struct snd_ca0106_details ca0106_chip_details[] = {
/* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
/* It is really just a normal SB Live 24bit. */
+ /* Tested:
+ * See ALSA bug#3251
+ */
+ { .serial = 0x10131102,
+ .name = "X-Fi Extreme Audio [SBxxxx]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+ /* It is really just a normal SB Live 24bit. */
/*
* CTRL:CA0111-WTLF
* ADC: WM8775SEDS
@@ -261,10 +274,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
/* hardware definition */
static struct snd_pcm_hardware snd_ca0106_playback_hw = {
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
.rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
SNDRV_PCM_RATE_192000),
@@ -447,6 +461,19 @@ static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime)
kfree(runtime->private_data);
}
+static const int spi_dacd_reg[] = {
+ [PCM_FRONT_CHANNEL] = SPI_DACD4_REG,
+ [PCM_REAR_CHANNEL] = SPI_DACD0_REG,
+ [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG,
+ [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG,
+};
+static const int spi_dacd_bit[] = {
+ [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT,
+ [PCM_REAR_CHANNEL] = SPI_DACD0_BIT,
+ [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT,
+ [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT,
+};
+
/* open_playback callback */
static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
int channel_id)
@@ -481,6 +508,17 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
return err;
if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
return err;
+ snd_pcm_set_sync(substream);
+
+ if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) {
+ const int reg = spi_dacd_reg[channel_id];
+
+ /* Power up dac */
+ chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id];
+ err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ if (err < 0)
+ return err;
+ }
return 0;
}
@@ -491,6 +529,14 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_ca0106_pcm *epcm = runtime->private_data;
chip->playback_channels[epcm->channel_id].use = 0;
+
+ if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) {
+ const int reg = spi_dacd_reg[epcm->channel_id];
+
+ /* Power down DAC */
+ chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id];
+ snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ }
/* FIXME: maybe zero others */
return 0;
}
@@ -809,6 +855,9 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
break;
}
snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
runtime = s->runtime;
epcm = runtime->private_data;
channel = epcm->channel_id;
@@ -1214,28 +1263,23 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s
return 0;
}
+#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value))
static unsigned int spi_dac_init[] = {
- 0x00ff,
- 0x02ff,
- 0x0400,
- 0x0520,
- 0x0620, /* Set 24 bit. Was 0x0600 */
- 0x08ff,
- 0x0aff,
- 0x0cff,
- 0x0eff,
- 0x10ff,
- 0x1200,
- 0x1400,
- 0x1480,
- 0x1800,
- 0x1aff,
- 0x1cff,
- 0x1e00,
- 0x0530,
- 0x0602,
- 0x0622,
- 0x1400,
+ SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */
+ SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT),
+ SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24),
+ SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB),
+ SPI_REG(9, 0x00),
+ SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT),
+ SPI_REG(12, 0x00),
+ SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE),
+ SPI_REG(SPI_DACD4_REG, 0x00),
};
static unsigned int i2c_adc_init[][2] = {
@@ -1475,8 +1519,13 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
int size, n;
size = ARRAY_SIZE(spi_dac_init);
- for (n=0; n < size; n++)
+ for (n = 0; n < size; n++) {
+ int reg = spi_dac_init[n] >> SPI_REG_SHIFT;
+
snd_ca0106_spi_write(chip, spi_dac_init[n]);
+ if (reg < ARRAY_SIZE(chip->spi_dac_reg))
+ chip->spi_dac_reg[reg] = spi_dac_init[n];
+ }
}
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 9c3a9c8d1dc2..be519a17dfa5 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.17
+ * Version: 0.0.18
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -39,6 +39,8 @@
* Modified Copyright message.
* 0.0.17
* Implement Mic and Line in Capture.
+ * 0.0.18
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
@@ -77,15 +79,7 @@
static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
-static int snd_ca0106_shared_spdif_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 snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info
static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -470,6 +464,42 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
return change;
}
+#define spi_mute_info snd_ctl_boolean_mono_info
+
+static int spi_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+
+ ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit);
+ return 0;
+}
+
+static int spi_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+ int ret;
+
+ ret = emu->spi_dac_reg[reg] & bit;
+ if (ucontrol->value.integer.value[0]) {
+ if (!ret) /* bit already cleared, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] &= ~bit;
+ } else {
+ if (ret) /* bit already set, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] |= bit;
+ }
+
+ ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]);
+ return ret ? -1 : 1;
+}
+
#define CA_VOLUME(xname,chid,reg) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -562,6 +592,28 @@ static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] __devinitdata =
I2C_VOLUME("Aux Capture Volume", 3),
};
+#define SPI_SWITCH(xname,reg,bit) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = spi_mute_info, \
+ .get = spi_mute_get, \
+ .put = spi_mute_put, \
+ .private_value = (reg<<SPI_REG_SHIFT) | (bit) \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_spi_dac_ctls[]
+__devinitdata = {
+ SPI_SWITCH("Analog Front Playback Switch",
+ SPI_DMUTE4_REG, SPI_DMUTE4_BIT),
+ SPI_SWITCH("Analog Rear Playback Switch",
+ SPI_DMUTE0_REG, SPI_DMUTE0_BIT),
+ SPI_SWITCH("Analog Center/LFE Playback Switch",
+ SPI_DMUTE2_REG, SPI_DMUTE2_BIT),
+ SPI_SWITCH("Analog Side Playback Switch",
+ SPI_DMUTE1_REG, SPI_DMUTE1_BIT),
+};
+
static int __devinit remove_ctl(struct snd_card *card, const char *name)
{
struct snd_ctl_elem_id id;
@@ -591,9 +643,19 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch
return -ENOENT;
}
+#define ADD_CTLS(emu, ctls) \
+ do { \
+ int i, err; \
+ for (i = 0; i < ARRAY_SIZE(ctls); i++) { \
+ err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
+ if (err < 0) \
+ return err; \
+ } \
+ } while (0)
+
int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
{
- int i, err;
+ int err;
struct snd_card *card = emu->card;
char **c;
static char *ca0106_remove_ctls[] = {
@@ -640,17 +702,9 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
rename_ctl(card, c[0], c[1]);
#endif
- for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_ctls); i++) {
- err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_ctls[i], emu));
- if (err < 0)
- return err;
- }
+ ADD_CTLS(emu, snd_ca0106_volume_ctls);
if (emu->details->i2c_adc == 1) {
- for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_i2c_adc_ctls); i++) {
- err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_i2c_adc_ctls[i], emu));
- if (err < 0)
- return err;
- }
+ ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls);
if (emu->details->gpio_type == 1)
err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
else /* gpio_type == 2 */
@@ -658,6 +712,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
if (err < 0)
return err;
}
+ if (emu->details->spi_dac == 1)
+ ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls);
return 0;
}
diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c
index 2e6eab1f1189..ad32eff2713f 100644
--- a/sound/pci/ca0106/ca_midi.c
+++ b/sound/pci/ca0106/ca_midi.c
@@ -6,7 +6,7 @@
* Changelog:
* Implementation is based on mpu401 and emu10k1x and
* tested with ca0106.
- * mpu401: Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h
index b72c0933bd22..922ed3e3731e 100644
--- a/sound/pci/ca0106/ca_midi.h
+++ b/sound/pci/ca0106/ca_midi.h
@@ -22,9 +22,9 @@
*
*/
-#include<linux/spinlock.h>
-#include<sound/rawmidi.h>
-#include<sound/mpu401.h>
+#include <linux/spinlock.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
#define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT
#define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 7d3c5ee0005c..6832649879ce 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -95,30 +95,34 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_CHADC0 0x00000001 /* ch0, 0:playback, 1:record */
#define CM_REG_FUNCTRL1 0x04
-#define CM_ASFC_MASK 0x0000E000 /* ADC sampling frequency */
-#define CM_ASFC_SHIFT 13
-#define CM_DSFC_MASK 0x00001C00 /* DAC sampling frequency */
-#define CM_DSFC_SHIFT 10
+#define CM_DSFC_MASK 0x0000E000 /* channel 1 (DAC?) sampling frequency */
+#define CM_DSFC_SHIFT 13
+#define CM_ASFC_MASK 0x00001C00 /* channel 0 (ADC?) sampling frequency */
+#define CM_ASFC_SHIFT 10
#define CM_SPDF_1 0x00000200 /* SPDIF IN/OUT at channel B */
#define CM_SPDF_0 0x00000100 /* SPDIF OUT only channel A */
-#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/OUT -> IN loopback */
+#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/IN -> OUT loopback */
#define CM_SPDO2DAC 0x00000040 /* SPDIF/OUT can be heard from internal DAC */
#define CM_INTRM 0x00000020 /* master control block (MCB) interrupt enabled */
#define CM_BREQ 0x00000010 /* bus master enabled */
#define CM_VOICE_EN 0x00000008 /* legacy voice (SB16,FM) */
-#define CM_UART_EN 0x00000004 /* UART */
-#define CM_JYSTK_EN 0x00000002 /* joy stick */
+#define CM_UART_EN 0x00000004 /* legacy UART */
+#define CM_JYSTK_EN 0x00000002 /* legacy joystick */
+#define CM_ZVPORT 0x00000001 /* ZVPORT */
#define CM_REG_CHFORMAT 0x08
#define CM_CHB3D5C 0x80000000 /* 5,6 channels */
+#define CM_FMOFFSET2 0x40000000 /* initial FM PCM offset 2 when Fmute=1 */
#define CM_CHB3D 0x20000000 /* 4 channels */
#define CM_CHIP_MASK1 0x1f000000
#define CM_CHIP_037 0x01000000
-
-#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */
+#define CM_SETLAT48 0x00800000 /* set latency timer 48h */
+#define CM_EDGEIRQ 0x00400000 /* emulated edge trigger legacy IRQ */
+#define CM_SPD24SEL39 0x00200000 /* 24-bit spdif: model 039 */
#define CM_AC3EN1 0x00100000 /* enable AC3: model 037 */
+#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */
#define CM_SPD24SEL 0x00020000 /* 24bit spdif: model 037 */
/* #define CM_SPDIF_INVERSE 0x00010000 */ /* ??? */
@@ -128,35 +132,45 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_ADCBITLEN_14 0x00008000
#define CM_ADCBITLEN_13 0x0000C000
-#define CM_ADCDACLEN_MASK 0x00003000
+#define CM_ADCDACLEN_MASK 0x00003000 /* model 037 */
#define CM_ADCDACLEN_060 0x00000000
#define CM_ADCDACLEN_066 0x00001000
#define CM_ADCDACLEN_130 0x00002000
#define CM_ADCDACLEN_280 0x00003000
+#define CM_ADCDLEN_MASK 0x00003000 /* model 039 */
+#define CM_ADCDLEN_ORIGINAL 0x00000000
+#define CM_ADCDLEN_EXTRA 0x00001000
+#define CM_ADCDLEN_24K 0x00002000
+#define CM_ADCDLEN_WEIGHT 0x00003000
+
#define CM_CH1_SRATE_176K 0x00000800
+#define CM_CH1_SRATE_96K 0x00000800 /* model 055? */
#define CM_CH1_SRATE_88K 0x00000400
#define CM_CH0_SRATE_176K 0x00000200
+#define CM_CH0_SRATE_96K 0x00000200 /* model 055? */
#define CM_CH0_SRATE_88K 0x00000100
#define CM_SPDIF_INVERSE2 0x00000080 /* model 055? */
+#define CM_DBLSPDS 0x00000040 /* double SPDIF sample rate 88.2/96 */
+#define CM_POLVALID 0x00000020 /* inverse SPDIF/IN valid bit */
+#define CM_SPDLOCKED 0x00000010
-#define CM_CH1FMT_MASK 0x0000000C
+#define CM_CH1FMT_MASK 0x0000000C /* bit 3: 16 bits, bit 2: stereo */
#define CM_CH1FMT_SHIFT 2
-#define CM_CH0FMT_MASK 0x00000003
+#define CM_CH0FMT_MASK 0x00000003 /* bit 1: 16 bits, bit 0: stereo */
#define CM_CH0FMT_SHIFT 0
#define CM_REG_INT_HLDCLR 0x0C
#define CM_CHIP_MASK2 0xff000000
+#define CM_CHIP_8768 0x20000000
+#define CM_CHIP_055 0x08000000
#define CM_CHIP_039 0x04000000
#define CM_CHIP_039_6CH 0x01000000
-#define CM_CHIP_055 0x08000000
-#define CM_CHIP_8768 0x20000000
+#define CM_UNKNOWN_INT_EN 0x00080000 /* ? */
#define CM_TDMA_INT_EN 0x00040000
#define CM_CH1_INT_EN 0x00020000
#define CM_CH0_INT_EN 0x00010000
-#define CM_INT_HOLD 0x00000002
-#define CM_INT_CLEAR 0x00000001
#define CM_REG_INT_STATUS 0x10
#define CM_INTR 0x80000000
@@ -175,12 +189,13 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_CHINT0 0x00000001
#define CM_REG_LEGACY_CTRL 0x14
-#define CM_NXCHG 0x80000000 /* h/w multi channels? */
+#define CM_NXCHG 0x80000000 /* don't map base reg dword->sample */
#define CM_VMPU_MASK 0x60000000 /* MPU401 i/o port address */
#define CM_VMPU_330 0x00000000
#define CM_VMPU_320 0x20000000
#define CM_VMPU_310 0x40000000
#define CM_VMPU_300 0x60000000
+#define CM_ENWR8237 0x10000000 /* enable bus master to write 8237 base reg */
#define CM_VSBSEL_MASK 0x0C000000 /* SB16 base address */
#define CM_VSBSEL_220 0x00000000
#define CM_VSBSEL_240 0x04000000
@@ -191,44 +206,74 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_FMSEL_3C8 0x01000000
#define CM_FMSEL_3E0 0x02000000
#define CM_FMSEL_3E8 0x03000000
-#define CM_ENSPDOUT 0x00800000 /* enable XPDIF/OUT to I/O interface */
-#define CM_SPDCOPYRHT 0x00400000 /* set copyright spdif in/out */
+#define CM_ENSPDOUT 0x00800000 /* enable XSPDIF/OUT to I/O interface */
+#define CM_SPDCOPYRHT 0x00400000 /* spdif in/out copyright bit */
#define CM_DAC2SPDO 0x00200000 /* enable wave+fm_midi -> SPDIF/OUT */
-#define CM_SETRETRY 0x00010000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */
+#define CM_INVIDWEN 0x00100000 /* internal vendor ID write enable, model 039? */
+#define CM_SETRETRY 0x00100000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */
+#define CM_C_EEACCESS 0x00080000 /* direct programming eeprom regs */
+#define CM_C_EECS 0x00040000
+#define CM_C_EEDI46 0x00020000
+#define CM_C_EECK46 0x00010000
#define CM_CHB3D6C 0x00008000 /* 5.1 channels support */
-#define CM_LINE_AS_BASS 0x00006000 /* use line-in as bass */
+#define CM_CENTR2LIN 0x00004000 /* line-in as center out */
+#define CM_BASE2LIN 0x00002000 /* line-in as bass out */
+#define CM_EXBASEN 0x00001000 /* external bass input enable */
#define CM_REG_MISC_CTRL 0x18
-#define CM_PWD 0x80000000
+#define CM_PWD 0x80000000 /* power down */
#define CM_RESET 0x40000000
-#define CM_SFIL_MASK 0x30000000
-#define CM_TXVX 0x08000000
-#define CM_N4SPK3D 0x04000000 /* 4ch output */
+#define CM_SFIL_MASK 0x30000000 /* filter control at front end DAC, model 037? */
+#define CM_VMGAIN 0x10000000 /* analog master amp +6dB, model 039? */
+#define CM_TXVX 0x08000000 /* model 037? */
+#define CM_N4SPK3D 0x04000000 /* copy front to rear */
#define CM_SPDO5V 0x02000000 /* 5V spdif output (1 = 0.5v (coax)) */
#define CM_SPDIF48K 0x01000000 /* write */
#define CM_SPATUS48K 0x01000000 /* read */
-#define CM_ENDBDAC 0x00800000 /* enable dual dac */
+#define CM_ENDBDAC 0x00800000 /* enable double dac */
#define CM_XCHGDAC 0x00400000 /* 0: front=ch0, 1: front=ch1 */
#define CM_SPD32SEL 0x00200000 /* 0: 16bit SPDIF, 1: 32bit */
-#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-IN -> int. OUT */
-#define CM_FM_EN 0x00080000 /* enalbe FM */
+#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-OUT -> int. IN */
+#define CM_FM_EN 0x00080000 /* enable legacy FM */
#define CM_AC3EN2 0x00040000 /* enable AC3: model 039 */
-#define CM_VIDWPDSB 0x00010000
+#define CM_ENWRASID 0x00010000 /* choose writable internal SUBID (audio) */
+#define CM_VIDWPDSB 0x00010000 /* model 037? */
#define CM_SPDF_AC97 0x00008000 /* 0: SPDIF/OUT 44.1K, 1: 48K */
-#define CM_MASK_EN 0x00004000
-#define CM_VIDWPPRT 0x00002000
-#define CM_SFILENB 0x00001000
-#define CM_MMODE_MASK 0x00000E00
+#define CM_MASK_EN 0x00004000 /* activate channel mask on legacy DMA */
+#define CM_ENWRMSID 0x00002000 /* choose writable internal SUBID (modem) */
+#define CM_VIDWPPRT 0x00002000 /* model 037? */
+#define CM_SFILENB 0x00001000 /* filter stepping at front end DAC, model 037? */
+#define CM_MMODE_MASK 0x00000E00 /* model DAA interface mode */
#define CM_SPDIF_SELECT2 0x00000100 /* for model > 039 ? */
#define CM_ENCENTER 0x00000080
-#define CM_FLINKON 0x00000040
-#define CM_FLINKOFF 0x00000020
-#define CM_MIDSMP 0x00000010
-#define CM_UPDDMA_MASK 0x0000000C
-#define CM_TWAIT_MASK 0x00000003
+#define CM_FLINKON 0x00000080 /* force modem link detection on, model 037 */
+#define CM_MUTECH1 0x00000040 /* mute PCI ch1 to DAC */
+#define CM_FLINKOFF 0x00000040 /* force modem link detection off, model 037 */
+#define CM_UNKNOWN_18_5 0x00000020 /* ? */
+#define CM_MIDSMP 0x00000010 /* 1/2 interpolation at front end DAC */
+#define CM_UPDDMA_MASK 0x0000000C /* TDMA position update notification */
+#define CM_UPDDMA_2048 0x00000000
+#define CM_UPDDMA_1024 0x00000004
+#define CM_UPDDMA_512 0x00000008
+#define CM_UPDDMA_256 0x0000000C
+#define CM_TWAIT_MASK 0x00000003 /* model 037 */
+#define CM_TWAIT1 0x00000002 /* FM i/o cycle, 0: 48, 1: 64 PCICLKs */
+#define CM_TWAIT0 0x00000001 /* i/o cycle, 0: 4, 1: 6 PCICLKs */
+
+#define CM_REG_TDMA_POSITION 0x1C
+#define CM_TDMA_CNT_MASK 0xFFFF0000 /* current byte/word count */
+#define CM_TDMA_ADR_MASK 0x0000FFFF /* current address */
/* byte */
#define CM_REG_MIXER0 0x20
+#define CM_REG_SBVR 0x20 /* write: sb16 version */
+#define CM_REG_DEV 0x20 /* read: hardware device version */
+
+#define CM_REG_MIXER21 0x21
+#define CM_UNKNOWN_21_MASK 0x78 /* ? */
+#define CM_X_ADPCM 0x04 /* SB16 ADPCM enable */
+#define CM_PROINV 0x02 /* SBPro left/right channel switching */
+#define CM_X_SB16 0x01 /* SB16 compatible */
#define CM_REG_SB16_DATA 0x22
#define CM_REG_SB16_ADDR 0x23
@@ -243,8 +288,8 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_FMMUTE_SHIFT 7
#define CM_WSMUTE 0x40 /* mute PCM */
#define CM_WSMUTE_SHIFT 6
-#define CM_SPK4 0x20 /* lin-in -> rear line out */
-#define CM_SPK4_SHIFT 5
+#define CM_REAR2LIN 0x20 /* lin-in -> rear line out */
+#define CM_REAR2LIN_SHIFT 5
#define CM_REAR2FRONT 0x10 /* exchange rear/front */
#define CM_REAR2FRONT_SHIFT 4
#define CM_WAVEINL 0x08 /* digital wave rec. left chan */
@@ -276,12 +321,13 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_VAUXR_MASK 0x0f
#define CM_REG_MISC 0x27
+#define CM_UNKNOWN_27_MASK 0xd8 /* ? */
#define CM_XGPO1 0x20
// #define CM_XGPBIO 0x04
#define CM_MIC_CENTER_LFE 0x04 /* mic as center/lfe out? (model 039 or later?) */
#define CM_SPDIF_INVERSE 0x04 /* spdif input phase inverse (model 037) */
#define CM_SPDVALID 0x02 /* spdif input valid check */
-#define CM_DMAUTO 0x01
+#define CM_DMAUTO 0x01 /* SB16 DMA auto detect */
#define CM_REG_AC97 0x28 /* hmmm.. do we have ac97 link? */
/*
@@ -322,18 +368,20 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
/*
* extended registers
*/
-#define CM_REG_CH0_FRAME1 0x80 /* base address */
-#define CM_REG_CH0_FRAME2 0x84
+#define CM_REG_CH0_FRAME1 0x80 /* write: base address */
+#define CM_REG_CH0_FRAME2 0x84 /* read: current address */
#define CM_REG_CH1_FRAME1 0x88 /* 0-15: count of samples at bus master; buffer size */
#define CM_REG_CH1_FRAME2 0x8C /* 16-31: count of samples at codec; fragment size */
+
#define CM_REG_EXT_MISC 0x90
-#define CM_REG_MISC_CTRL_8768 0x92 /* reg. name the same as 0x18 */
-#define CM_CHB3D8C 0x20 /* 7.1 channels support */
-#define CM_SPD32FMT 0x10 /* SPDIF/IN 32k */
-#define CM_ADC2SPDIF 0x08 /* ADC output to SPDIF/OUT */
-#define CM_SHAREADC 0x04 /* DAC in ADC as Center/LFE */
-#define CM_REALTCMP 0x02 /* monitor the CMPL/CMPR of ADC */
-#define CM_INVLRCK 0x01 /* invert ZVPORT's LRCK */
+#define CM_ADC48K44K 0x10000000 /* ADC parameters group, 0: 44k, 1: 48k */
+#define CM_CHB3D8C 0x00200000 /* 7.1 channels support */
+#define CM_SPD32FMT 0x00100000 /* SPDIF/IN 32k sample rate */
+#define CM_ADC2SPDIF 0x00080000 /* ADC output to SPDIF/OUT */
+#define CM_SHAREADC 0x00040000 /* DAC in ADC as Center/LFE */
+#define CM_REALTCMP 0x00020000 /* monitor the CMPL/CMPR of ADC */
+#define CM_INVLRCK 0x00010000 /* invert ZVPORT's LRCK */
+#define CM_UNKNOWN_90_MASK 0x0000FFFF /* ? */
/*
* size of i/o region
@@ -383,15 +431,14 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
struct cmipci_pcm {
struct snd_pcm_substream *substream;
- int running; /* dac/adc running? */
+ u8 running; /* dac/adc running? */
+ u8 fmt; /* format bits */
+ u8 is_dac;
+ u8 needs_silencing;
unsigned int dma_size; /* in frames */
- unsigned int period_size; /* in frames */
+ unsigned int shift;
+ unsigned int ch; /* channel (0/1) */
unsigned int offset; /* physical address of the buffer */
- unsigned int fmt; /* format bits */
- int ch; /* channel (0/1) */
- unsigned int is_dac; /* is dac? */
- int bytes_per_frame;
- int shift;
};
/* mixer elements toggled/resumed during ac3 playback */
@@ -424,7 +471,6 @@ struct cmipci {
int chip_version;
int max_channels;
- unsigned int has_dual_dac: 1;
unsigned int can_ac3_sw: 1;
unsigned int can_ac3_hw: 1;
unsigned int can_multi_ch: 1;
@@ -557,6 +603,9 @@ static unsigned int rates[] = { 5512, 11025, 22050, 44100, 8000, 16000, 32000, 4
static unsigned int snd_cmipci_rate_freq(unsigned int rate)
{
unsigned int i;
+
+ if (rate > 48000)
+ rate /= 2;
for (i = 0; i < ARRAY_SIZE(rates); i++) {
if (rates[i] == rate)
return i;
@@ -671,19 +720,19 @@ static int snd_cmipci_hw_free(struct snd_pcm_substream *substream)
/*
*/
-static unsigned int hw_channels[] = {1, 2, 4, 5, 6, 8};
+static unsigned int hw_channels[] = {1, 2, 4, 6, 8};
static struct snd_pcm_hw_constraint_list hw_constraints_channels_4 = {
.count = 3,
.list = hw_channels,
.mask = 0,
};
static struct snd_pcm_hw_constraint_list hw_constraints_channels_6 = {
- .count = 5,
+ .count = 4,
.list = hw_channels,
.mask = 0,
};
static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = {
- .count = 6,
+ .count = 5,
.list = hw_channels,
.mask = 0,
};
@@ -691,48 +740,37 @@ static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = {
static int set_dac_channels(struct cmipci *cm, struct cmipci_pcm *rec, int channels)
{
if (channels > 2) {
- if (! cm->can_multi_ch)
+ if (!cm->can_multi_ch || !rec->ch)
return -EINVAL;
if (rec->fmt != 0x03) /* stereo 16bit only */
return -EINVAL;
+ }
+ if (cm->can_multi_ch) {
spin_lock_irq(&cm->reg_lock);
- snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
- snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
- if (channels > 4) {
- snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
- snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
+ if (channels > 2) {
+ snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
} else {
- snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
- snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
}
- if (channels >= 6) {
+ if (channels == 8)
+ snd_cmipci_set_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C);
+ if (channels == 6) {
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
- snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER);
} else {
- snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER);
- }
- if (cm->chip_version == 68) {
- if (channels == 8) {
- snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL_8768, CM_CHB3D8C);
- } else {
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL_8768, CM_CHB3D8C);
- }
- }
- spin_unlock_irq(&cm->reg_lock);
-
- } else {
- if (cm->can_multi_ch) {
- spin_lock_irq(&cm->reg_lock);
- snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
- snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER);
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
- spin_unlock_irq(&cm->reg_lock);
}
+ if (channels == 4)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ spin_unlock_irq(&cm->reg_lock);
}
return 0;
}
@@ -746,6 +784,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
struct snd_pcm_substream *substream)
{
unsigned int reg, freq, val;
+ unsigned int period_size;
struct snd_pcm_runtime *runtime = substream->runtime;
rec->fmt = 0;
@@ -765,11 +804,11 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
rec->offset = runtime->dma_addr;
/* buffer and period sizes in frame */
rec->dma_size = runtime->buffer_size << rec->shift;
- rec->period_size = runtime->period_size << rec->shift;
+ period_size = runtime->period_size << rec->shift;
if (runtime->channels > 2) {
/* multi-channels */
rec->dma_size = (rec->dma_size * runtime->channels) / 2;
- rec->period_size = (rec->period_size * runtime->channels) / 2;
+ period_size = (period_size * runtime->channels) / 2;
}
spin_lock_irq(&cm->reg_lock);
@@ -780,7 +819,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
/* program sample counts */
reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
snd_cmipci_write_w(cm, reg, rec->dma_size - 1);
- snd_cmipci_write_w(cm, reg + 2, rec->period_size - 1);
+ snd_cmipci_write_w(cm, reg + 2, period_size - 1);
/* set adc/dac flag */
val = rec->ch ? CM_CHADC1 : CM_CHADC0;
@@ -795,11 +834,11 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
freq = snd_cmipci_rate_freq(runtime->rate);
val = snd_cmipci_read(cm, CM_REG_FUNCTRL1);
if (rec->ch) {
- val &= ~CM_ASFC_MASK;
- val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK;
- } else {
val &= ~CM_DSFC_MASK;
val |= (freq << CM_DSFC_SHIFT) & CM_DSFC_MASK;
+ } else {
+ val &= ~CM_ASFC_MASK;
+ val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK;
}
snd_cmipci_write(cm, CM_REG_FUNCTRL1, val);
//snd_printd("cmipci: functrl1 = %08x\n", val);
@@ -813,6 +852,16 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
val &= ~CM_CH0FMT_MASK;
val |= rec->fmt << CM_CH0FMT_SHIFT;
}
+ if (cm->chip_version == 68) {
+ if (runtime->rate == 88200)
+ val |= CM_CH0_SRATE_88K << (rec->ch * 2);
+ else
+ val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2));
+ if (runtime->rate == 96000)
+ val |= CM_CH0_SRATE_96K << (rec->ch * 2);
+ else
+ val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2));
+ }
snd_cmipci_write(cm, CM_REG_CHFORMAT, val);
//snd_printd("cmipci: chformat = %08x\n", val);
@@ -826,7 +875,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
* PCM trigger/stop
*/
static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec,
- struct snd_pcm_substream *substream, int cmd)
+ int cmd)
{
unsigned int inthld, chen, reset, pause;
int result = 0;
@@ -855,6 +904,7 @@ static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec,
cm->ctrl &= ~chen;
snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | reset);
snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~reset);
+ rec->needs_silencing = rec->is_dac;
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
@@ -906,7 +956,7 @@ static int snd_cmipci_playback_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct cmipci *cm = snd_pcm_substream_chip(substream);
- return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], substream, cmd);
+ return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], cmd);
}
static snd_pcm_uframes_t snd_cmipci_playback_pointer(struct snd_pcm_substream *substream)
@@ -925,7 +975,7 @@ static int snd_cmipci_capture_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct cmipci *cm = snd_pcm_substream_chip(substream);
- return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], substream, cmd);
+ return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], cmd);
}
static snd_pcm_uframes_t snd_cmipci_capture_pointer(struct snd_pcm_substream *substream)
@@ -1199,15 +1249,19 @@ static int setup_spdif_playback(struct cmipci *cm, struct snd_pcm_substream *sub
snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
setup_ac3(cm, subs, do_ac3, rate);
- if (rate == 48000)
+ if (rate == 48000 || rate == 96000)
snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97);
else
snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97);
-
+ if (rate > 48000)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
} else {
/* they are controlled via "IEC958 Output Switch" */
/* snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_ENSPDOUT); */
/* snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_SPDO2DAC); */
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
setup_ac3(cm, subs, 0, 0);
}
@@ -1227,7 +1281,7 @@ static int snd_cmipci_playback_prepare(struct snd_pcm_substream *substream)
int rate = substream->runtime->rate;
int err, do_spdif, do_ac3 = 0;
- do_spdif = ((rate == 44100 || rate == 48000) &&
+ do_spdif = (rate >= 44100 &&
substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE &&
substream->runtime->channels == 2);
if (do_spdif && cm->can_ac3_hw)
@@ -1252,11 +1306,75 @@ static int snd_cmipci_playback_spdif_prepare(struct snd_pcm_substream *substream
return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_PLAY], substream);
}
+/*
+ * Apparently, the samples last played on channel A stay in some buffer, even
+ * after the channel is reset, and get added to the data for the rear DACs when
+ * playing a multichannel stream on channel B. This is likely to generate
+ * wraparounds and thus distortions.
+ * To avoid this, we play at least one zero sample after the actual stream has
+ * stopped.
+ */
+static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec)
+{
+ struct snd_pcm_runtime *runtime = rec->substream->runtime;
+ unsigned int reg, val;
+
+ if (rec->needs_silencing && runtime && runtime->dma_area) {
+ /* set up a small silence buffer */
+ memset(runtime->dma_area, 0, PAGE_SIZE);
+ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
+ val = ((PAGE_SIZE / 4) - 1) | (((PAGE_SIZE / 4) / 2 - 1) << 16);
+ snd_cmipci_write(cm, reg, val);
+
+ /* configure for 16 bits, 2 channels, 8 kHz */
+ if (runtime->channels > 2)
+ set_dac_channels(cm, rec, 2);
+ spin_lock_irq(&cm->reg_lock);
+ val = snd_cmipci_read(cm, CM_REG_FUNCTRL1);
+ val &= ~(CM_ASFC_MASK << (rec->ch * 3));
+ val |= (4 << CM_ASFC_SHIFT) << (rec->ch * 3);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL1, val);
+ val = snd_cmipci_read(cm, CM_REG_CHFORMAT);
+ val &= ~(CM_CH0FMT_MASK << (rec->ch * 2));
+ val |= (3 << CM_CH0FMT_SHIFT) << (rec->ch * 2);
+ if (cm->chip_version == 68) {
+ val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2));
+ val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2));
+ }
+ snd_cmipci_write(cm, CM_REG_CHFORMAT, val);
+
+ /* start stream (we don't need interrupts) */
+ cm->ctrl |= CM_CHEN0 << rec->ch;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ spin_unlock_irq(&cm->reg_lock);
+
+ msleep(1);
+
+ /* stop and reset stream */
+ spin_lock_irq(&cm->reg_lock);
+ cm->ctrl &= ~(CM_CHEN0 << rec->ch);
+ val = CM_RST_CH0 << rec->ch;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | val);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~val);
+ spin_unlock_irq(&cm->reg_lock);
+
+ rec->needs_silencing = 0;
+ }
+}
+
static int snd_cmipci_playback_hw_free(struct snd_pcm_substream *substream)
{
struct cmipci *cm = snd_pcm_substream_chip(substream);
setup_spdif_playback(cm, substream, 0, 0);
restore_mixer_state(cm);
+ snd_cmipci_silence_hack(cm, &cm->channel[0]);
+ return snd_cmipci_hw_free(substream);
+}
+
+static int snd_cmipci_playback2_hw_free(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ snd_cmipci_silence_hack(cm, &cm->channel[1]);
return snd_cmipci_hw_free(substream);
}
@@ -1515,7 +1633,11 @@ static int snd_cmipci_playback_open(struct snd_pcm_substream *substream)
if ((err = open_device_check(cm, CM_OPEN_PLAYBACK, substream)) < 0)
return err;
runtime->hw = snd_cmipci_playback;
- runtime->hw.channels_max = cm->max_channels;
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
cm->dig_pcm_status = cm->dig_status;
return 0;
@@ -1558,9 +1680,14 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream)
else if (cm->max_channels == 8)
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_8);
}
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
}
mutex_unlock(&cm->open_mutex);
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
return 0;
}
@@ -1574,8 +1701,15 @@ static int snd_cmipci_playback_spdif_open(struct snd_pcm_substream *substream)
return err;
if (cm->can_ac3_hw) {
runtime->hw = snd_cmipci_playback_spdif;
- if (cm->chip_version >= 37)
+ if (cm->chip_version >= 37) {
runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ }
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
} else {
runtime->hw = snd_cmipci_playback_iec958_subframe;
}
@@ -1668,7 +1802,7 @@ static struct snd_pcm_ops snd_cmipci_playback2_ops = {
.close = snd_cmipci_playback2_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_cmipci_playback2_hw_params,
- .hw_free = snd_cmipci_hw_free,
+ .hw_free = snd_cmipci_playback2_hw_free,
.prepare = snd_cmipci_capture_prepare, /* channel B */
.trigger = snd_cmipci_capture_trigger, /* channel B */
.pointer = snd_cmipci_capture_pointer, /* channel B */
@@ -2139,15 +2273,7 @@ struct cmipci_switch_args {
*/
};
-static int snd_cmipci_uswitch_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 snd_cmipci_uswitch_info snd_ctl_boolean_mono_info
static int _snd_cmipci_uswitch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
@@ -2260,8 +2386,8 @@ DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, 0, 0, 0); /* rever
DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, CM_XCHGDAC, 0, 0);
#endif
DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MISC_CTRL, CM_N4SPK3D, 0, 0);
-// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_SPK4, 1, 0);
-// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS, 0, 0);
+// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0);
+// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0);
// DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */
DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0);
@@ -2331,11 +2457,11 @@ static inline unsigned int get_line_in_mode(struct cmipci *cm)
unsigned int val;
if (cm->chip_version >= 39) {
val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL);
- if (val & CM_LINE_AS_BASS)
+ if (val & (CM_CENTR2LIN | CM_BASE2LIN))
return 2;
}
val = snd_cmipci_read_b(cm, CM_REG_MIXER1);
- if (val & CM_SPK4)
+ if (val & CM_REAR2LIN)
return 1;
return 0;
}
@@ -2359,13 +2485,13 @@ static int snd_cmipci_line_in_mode_put(struct snd_kcontrol *kcontrol,
spin_lock_irq(&cm->reg_lock);
if (ucontrol->value.enumerated.item[0] == 2)
- change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS);
+ change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN);
else
- change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS);
+ change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN);
if (ucontrol->value.enumerated.item[0] == 1)
- change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_SPK4);
+ change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN);
else
- change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_SPK4);
+ change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN);
spin_unlock_irq(&cm->reg_lock);
return change;
}
@@ -2583,19 +2709,18 @@ static void snd_cmipci_proc_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct cmipci *cm = entry->private_data;
- int i;
+ int i, v;
- snd_iprintf(buffer, "%s\n\n", cm->card->longname);
- for (i = 0; i < 0x40; i++) {
- int v = inb(cm->iobase + i);
+ snd_iprintf(buffer, "%s\n", cm->card->longname);
+ for (i = 0; i < 0x94; i++) {
+ if (i == 0x28)
+ i = 0x90;
+ v = inb(cm->iobase + i);
if (i % 4 == 0)
- snd_iprintf(buffer, "%02x: ", i);
- snd_iprintf(buffer, "%02x", v);
- if (i % 4 == 3)
- snd_iprintf(buffer, "\n");
- else
- snd_iprintf(buffer, " ");
+ snd_iprintf(buffer, "\n%02x:", i);
+ snd_iprintf(buffer, " %02x", v);
}
+ snd_iprintf(buffer, "\n");
}
static void __devinit snd_cmipci_proc_init(struct cmipci *cm)
@@ -2633,46 +2758,40 @@ static void __devinit query_chip(struct cmipci *cm)
if (! detect) {
/* check reg 08h, bit 24-28 */
detect = snd_cmipci_read(cm, CM_REG_CHFORMAT) & CM_CHIP_MASK1;
- if (! detect) {
+ switch (detect) {
+ case 0:
cm->chip_version = 33;
- cm->max_channels = 2;
if (cm->do_soft_ac3)
cm->can_ac3_sw = 1;
else
cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- } else {
+ break;
+ case CM_CHIP_037:
cm->chip_version = 37;
- cm->max_channels = 2;
cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
+ break;
+ default:
+ cm->chip_version = 39;
+ cm->can_ac3_hw = 1;
+ break;
}
+ cm->max_channels = 2;
} else {
- /* check reg 0Ch, bit 26 */
- if (detect & CM_CHIP_8768) {
- cm->chip_version = 68;
- cm->max_channels = 8;
- cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- cm->can_multi_ch = 1;
- } else if (detect & CM_CHIP_055) {
- cm->chip_version = 55;
- cm->max_channels = 6;
- cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- cm->can_multi_ch = 1;
- } else if (detect & CM_CHIP_039) {
+ if (detect & CM_CHIP_039) {
cm->chip_version = 39;
if (detect & CM_CHIP_039_6CH) /* 4 or 6 channels */
cm->max_channels = 6;
else
cm->max_channels = 4;
- cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- cm->can_multi_ch = 1;
+ } else if (detect & CM_CHIP_8768) {
+ cm->chip_version = 68;
+ cm->max_channels = 8;
} else {
- printk(KERN_ERR "chip %x version not supported\n", detect);
+ cm->chip_version = 55;
+ cm->max_channels = 6;
}
+ cm->can_ac3_hw = 1;
+ cm->can_multi_ch = 1;
}
}
@@ -2782,10 +2901,14 @@ static int __devinit snd_cmipci_create_fm(struct cmipci *cm, long fm_port)
if (!fm_port)
goto disable_fm;
- /* first try FM regs in PCI port range */
- iosynth = cm->iobase + CM_REG_FM_PCI;
- err = snd_opl3_create(cm->card, iosynth, iosynth + 2,
- OPL3_HW_OPL3, 1, &opl3);
+ if (cm->chip_version >= 39) {
+ /* first try FM regs in PCI port range */
+ iosynth = cm->iobase + CM_REG_FM_PCI;
+ err = snd_opl3_create(cm->card, iosynth, iosynth + 2,
+ OPL3_HW_OPL3, 1, &opl3);
+ } else {
+ err = -EIO;
+ }
if (err < 0) {
/* then try legacy ports */
val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK;
@@ -2829,9 +2952,10 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
static struct snd_device_ops ops = {
.dev_free = snd_cmipci_dev_free,
};
- unsigned int val = 0;
+ unsigned int val;
long iomidi;
- int integrated_midi;
+ int integrated_midi = 0;
+ char modelstr[16];
int pcm_index, pcm_spdif_index;
static struct pci_device_id intel_82437vx[] = {
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) },
@@ -2904,6 +3028,8 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
#endif
/* initialize codec registers */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_RESET);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_RESET);
snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */
snd_cmipci_ch_reset(cm, CM_CH_PLAY);
snd_cmipci_ch_reset(cm, CM_CH_CAPT);
@@ -2917,6 +3043,10 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
#else
snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
#endif
+ if (cm->chip_version) {
+ snd_cmipci_write_b(cm, CM_REG_EXT_MISC, 0x20); /* magic */
+ snd_cmipci_write_b(cm, CM_REG_EXT_MISC + 1, 0x09); /* more magic */
+ }
/* Set Bus Master Request */
snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_BREQ);
@@ -2931,15 +3061,55 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
break;
}
+ if (cm->chip_version < 68) {
+ val = pci->device < 0x110 ? 8338 : 8738;
+ } else {
+ switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) {
+ case 0:
+ val = 8769;
+ break;
+ case 2:
+ val = 8762;
+ break;
+ default:
+ switch ((pci->subsystem_vendor << 16) |
+ pci->subsystem_device) {
+ case 0x13f69761:
+ case 0x584d3741:
+ case 0x584d3751:
+ case 0x584d3761:
+ case 0x584d3771:
+ case 0x72848384:
+ val = 8770;
+ break;
+ default:
+ val = 8768;
+ break;
+ }
+ }
+ }
+ sprintf(card->shortname, "C-Media CMI%d", val);
+ if (cm->chip_version < 68)
+ sprintf(modelstr, " (model %d)", cm->chip_version);
+ else
+ modelstr[0] = '\0';
+ sprintf(card->longname, "%s%s at %#lx, irq %i",
+ card->shortname, modelstr, cm->iobase, cm->irq);
+
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, cm, &ops)) < 0) {
snd_cmipci_free(cm);
return err;
}
- integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff;
- if (integrated_midi && mpu_port[dev] == 1)
- iomidi = cm->iobase + CM_REG_MPU_PCI;
- else {
+ if (cm->chip_version >= 39) {
+ val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1);
+ if (val != 0x00 && val != 0xff) {
+ iomidi = cm->iobase + CM_REG_MPU_PCI;
+ integrated_midi = 1;
+ }
+ }
+ if (!integrated_midi) {
+ val = 0;
iomidi = mpu_port[dev];
switch (iomidi) {
case 0x320: val = CM_VMPU_320; break;
@@ -2953,11 +3123,21 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
/* enable UART */
snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN);
+ if (inb(iomidi + 1) == 0xff) {
+ snd_printk(KERN_ERR "cannot enable MPU-401 port"
+ " at %#lx\n", iomidi);
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1,
+ CM_UART_EN);
+ iomidi = 0;
+ }
}
}
- if ((err = snd_cmipci_create_fm(cm, fm_port[dev])) < 0)
- return err;
+ if (cm->chip_version < 68) {
+ err = snd_cmipci_create_fm(cm, fm_port[dev]);
+ if (err < 0)
+ return err;
+ }
/* reset mixer */
snd_cmipci_mixer_write(cm, 0, 0);
@@ -2969,11 +3149,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
if ((err = snd_cmipci_pcm_new(cm, pcm_index)) < 0)
return err;
pcm_index++;
- if (cm->has_dual_dac) {
- if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0)
- return err;
- pcm_index++;
- }
+ if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0)
+ return err;
+ pcm_index++;
if (cm->can_ac3_hw || cm->can_ac3_sw) {
pcm_spdif_index = pcm_index;
if ((err = snd_cmipci_pcm_spdif_new(cm, pcm_index)) < 0)
@@ -3057,15 +3235,6 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci,
}
card->private_data = cm;
- sprintf(card->shortname, "C-Media PCI %s", card->driver);
- sprintf(card->longname, "%s (model %d) at 0x%lx, irq %i",
- card->shortname,
- cm->chip_version,
- cm->iobase,
- cm->irq);
-
- //snd_printd("%s is detected\n", card->longname);
-
if ((err = snd_card_register(card)) < 0) {
snd_card_free(card);
return err;
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 44cf54607647..9a55f4a9739b 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1,6 +1,6 @@
/*
* Driver for Cirrus Logic CS4281 based PCI soundcard
- * 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
@@ -38,7 +38,7 @@
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Cirrus Logic CS4281");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}");
@@ -842,12 +842,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream)
static struct snd_pcm_hardware snd_cs4281_playback =
{
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_SYNC_START),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
@@ -868,12 +867,11 @@ static struct snd_pcm_hardware snd_cs4281_playback =
static struct snd_pcm_hardware snd_cs4281_capture =
{
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_SYNC_START),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
@@ -904,7 +902,6 @@ static int snd_cs4281_playback_open(struct snd_pcm_substream *substream)
dma->right_slot = 1;
runtime->private_data = dma;
runtime->hw = snd_cs4281_playback;
- snd_pcm_set_sync(substream);
/* should be detected from the AC'97 layer, but it seems
that although CS4297A rev B reports 18-bit ADC resolution,
samples are 20-bit */
@@ -924,7 +921,6 @@ static int snd_cs4281_capture_open(struct snd_pcm_substream *substream)
dma->right_slot = 11;
runtime->private_data = dma;
runtime->hw = snd_cs4281_capture;
- snd_pcm_set_sync(substream);
/* should be detected from the AC'97 layer, but it seems
that although CS4297A rev B reports 18-bit ADC resolution,
samples are 20-bit */
diff --git a/sound/pci/cs46xx/Makefile b/sound/pci/cs46xx/Makefile
index d8b77b89aec4..67e811ec8539 100644
--- a/sound/pci/cs46xx/Makefile
+++ b/sound/pci/cs46xx/Makefile
@@ -1,12 +1,10 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-cs46xx-objs := cs46xx.o cs46xx_lib.o
-ifeq ($(CONFIG_SND_CS46XX_NEW_DSP),y)
- snd-cs46xx-objs += dsp_spos.o dsp_spos_scb_lib.o
-endif
+snd-cs46xx-y := cs46xx.o cs46xx_lib.o
+snd-cs46xx-$(CONFIG_SND_CS46XX_NEW_DSP) += dsp_spos.o dsp_spos_scb_lib.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS46XX) += snd-cs46xx.o
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 8b6cd144d101..2699cb6c2cd6 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -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
@@ -34,7 +34,7 @@
#include <sound/cs46xx.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)},"
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 71d7aab9d869..2c7bfc9fef61 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Abramo Bagnara <abramo@alsa-project.org>
* Cirrus Logic, Inc.
* Routines for control of Cirrus Logic CS461x chips
@@ -1818,15 +1818,7 @@ static int snd_cs46xx_vol_iec958_put(struct snd_kcontrol *kcontrol, struct snd_c
}
#endif
-static int snd_mixer_boolean_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 snd_mixer_boolean_info snd_ctl_boolean_mono_info
static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
index 20dcd72f06c1..018a7de56017 100644
--- a/sound/pci/cs46xx/cs46xx_lib.h
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -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
diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h
index 0d246bca4184..f9e169d33c03 100644
--- a/sound/pci/cs46xx/dsp_spos.h
+++ b/sound/pci/cs46xx/dsp_spos.h
@@ -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
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 57e357de1500..eded4dfeba12 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1480,7 +1480,7 @@ void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip,
if (!pcm_channel->src_scb->ref_count) {
cs46xx_dsp_remove_scb(chip,pcm_channel->src_scb);
- snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot <= DSP_MAX_SRC_NR,
+ snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot < DSP_MAX_SRC_NR,
return );
ins->src_scb_slots[pcm_channel->src_slot] = 0;
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index ad947b4c04cc..bb3d57e6a3cb 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -2,11 +2,8 @@
# Makefile for cs5535audio
#
-snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o
-
-ifeq ($(CONFIG_PM),y)
-snd-cs5535audio-objs += cs5535audio_pm.o
-endif
+snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
+snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index b8e75ef9c1e6..2b35889787be 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -206,7 +206,6 @@ static void process_bm1_irq(struct cs5535audio *cs5535au)
static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
{
u16 acc_irq_stat;
- u8 bm_stat;
unsigned char count;
struct cs5535audio *cs5535au = dev_id;
@@ -217,7 +216,7 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
if (!acc_irq_stat)
return IRQ_NONE;
- for (count = 0; count < 10; count++) {
+ for (count = 0; count < 4; count++) {
if (acc_irq_stat & (1 << count)) {
switch (count) {
case IRQ_STS:
@@ -232,26 +231,9 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
case BM1_IRQ_STS:
process_bm1_irq(cs5535au);
break;
- case BM2_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM2_STATUS);
- break;
- case BM3_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM3_STATUS);
- break;
- case BM4_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM4_STATUS);
- break;
- case BM5_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM5_STATUS);
- break;
- case BM6_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM6_STATUS);
- break;
- case BM7_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM7_STATUS);
- break;
default:
- snd_printk(KERN_ERR "Unexpected irq src\n");
+ snd_printk(KERN_ERR "Unexpected irq src: "
+ "0x%x\n", acc_irq_stat);
break;
}
}
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 4fd1f31a6cf9..66bae7664193 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -16,57 +16,28 @@
#define ACC_IRQ_STATUS 0x12
#define ACC_BM0_CMD 0x20
#define ACC_BM1_CMD 0x28
-#define ACC_BM2_CMD 0x30
-#define ACC_BM3_CMD 0x38
-#define ACC_BM4_CMD 0x40
-#define ACC_BM5_CMD 0x48
-#define ACC_BM6_CMD 0x50
-#define ACC_BM7_CMD 0x58
#define ACC_BM0_PRD 0x24
#define ACC_BM1_PRD 0x2C
-#define ACC_BM2_PRD 0x34
-#define ACC_BM3_PRD 0x3C
-#define ACC_BM4_PRD 0x44
-#define ACC_BM5_PRD 0x4C
-#define ACC_BM6_PRD 0x54
-#define ACC_BM7_PRD 0x5C
#define ACC_BM0_STATUS 0x21
#define ACC_BM1_STATUS 0x29
-#define ACC_BM2_STATUS 0x31
-#define ACC_BM3_STATUS 0x39
-#define ACC_BM4_STATUS 0x41
-#define ACC_BM5_STATUS 0x49
-#define ACC_BM6_STATUS 0x51
-#define ACC_BM7_STATUS 0x59
#define ACC_BM0_PNTR 0x60
#define ACC_BM1_PNTR 0x64
-#define ACC_BM2_PNTR 0x68
-#define ACC_BM3_PNTR 0x6C
-#define ACC_BM4_PNTR 0x70
-#define ACC_BM5_PNTR 0x74
-#define ACC_BM6_PNTR 0x78
-#define ACC_BM7_PNTR 0x7C
+
/* acc_codec bar0 reg bits */
/* ACC_IRQ_STATUS */
#define IRQ_STS 0
#define WU_IRQ_STS 1
#define BM0_IRQ_STS 2
#define BM1_IRQ_STS 3
-#define BM2_IRQ_STS 4
-#define BM3_IRQ_STS 5
-#define BM4_IRQ_STS 6
-#define BM5_IRQ_STS 7
-#define BM6_IRQ_STS 8
-#define BM7_IRQ_STS 9
/* ACC_BMX_STATUS */
#define EOP (1<<0)
#define BM_EOP_ERR (1<<1)
/* ACC_BMX_CTL */
-#define BM_CTL_EN 0x00000001
-#define BM_CTL_PAUSE 0x00000011
-#define BM_CTL_DIS 0x00000000
-#define BM_CTL_BYTE_ORD_LE 0x00000000
-#define BM_CTL_BYTE_ORD_BE 0x00000100
+#define BM_CTL_EN 0x01
+#define BM_CTL_PAUSE 0x03
+#define BM_CTL_DIS 0x00
+#define BM_CTL_BYTE_ORD_LE 0x00
+#define BM_CTL_BYTE_ORD_BE 0x04
/* cs5535 specific ac97 codec register defines */
#define CMD_MASK 0xFF00FFFF
#define CMD_NEW 0x00010000
@@ -106,7 +77,6 @@ struct cs5535audio_dma {
struct snd_pcm_substream *substream;
unsigned int buf_addr, buf_bytes;
unsigned int period_bytes, periods;
- int suspended;
u32 saved_prd;
};
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index 5450a9e8f133..21df0634af32 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -43,7 +43,6 @@ static struct snd_pcm_hardware snd_cs5535audio_playback =
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_SYNC_START |
SNDRV_PCM_INFO_RESUME
),
.formats = (
@@ -71,8 +70,7 @@ static struct snd_pcm_hardware snd_cs5535audio_capture =
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_SYNC_START
+ SNDRV_PCM_INFO_MMAP_VALID
),
.formats = (
SNDRV_PCM_FMTBIT_S16_LE
@@ -102,7 +100,6 @@ static int snd_cs5535audio_playback_open(struct snd_pcm_substream *substream)
runtime->hw = snd_cs5535audio_playback;
cs5535au->playback_substream = substream;
runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK]);
- snd_pcm_set_sync(substream);
if ((err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
@@ -164,6 +161,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
jmpprd_addr = cpu_to_le32(lastdesc->addr +
(sizeof(struct cs5535audio_dma_desc)*periods));
+ dma->substream = substream;
dma->period_bytes = period_bytes;
dma->periods = periods;
spin_lock_irq(&cs5535au->reg_lock);
@@ -241,6 +239,7 @@ static void cs5535audio_clear_dma_packets(struct cs5535audio *cs5535au,
{
snd_dma_free_pages(&dma->desc_buf);
dma->desc_buf.area = NULL;
+ dma->substream = NULL;
}
static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream,
@@ -298,14 +297,12 @@ static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd)
break;
case SNDRV_PCM_TRIGGER_RESUME:
dma->ops->enable_dma(cs5535au);
- dma->suspended = 0;
break;
case SNDRV_PCM_TRIGGER_STOP:
dma->ops->disable_dma(cs5535au);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
dma->ops->disable_dma(cs5535au);
- dma->suspended = 1;
break;
default:
snd_printk(KERN_ERR "unhandled trigger\n");
@@ -348,7 +345,6 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream)
runtime->hw = snd_cs5535audio_capture;
cs5535au->capture_substream = substream;
runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE]);
- snd_pcm_set_sync(substream);
if ((err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index 3e4d198a4502..838708f6d45e 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -64,18 +64,21 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
int i;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(cs5535au->pcm);
+ snd_ac97_suspend(cs5535au->ac97);
for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
struct cs5535audio_dma *dma = &cs5535au->dmas[i];
- if (dma && dma->substream && !dma->suspended)
+ if (dma && dma->substream)
dma->saved_prd = dma->ops->read_prd(cs5535au);
}
- snd_pcm_suspend_all(cs5535au->pcm);
- snd_ac97_suspend(cs5535au->ac97);
/* save important regs, then disable aclink in hw */
snd_cs5535audio_stop_hardware(cs5535au);
+ if (pci_save_state(pci)) {
+ printk(KERN_ERR "cs5535audio: pci_save_state failed!\n");
+ return -EIO;
+ }
pci_disable_device(pci);
- pci_save_state(pci);
pci_set_power_state(pci, pci_choose_state(pci, state));
return 0;
}
@@ -89,7 +92,12 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
int i;
pci_set_power_state(pci, PCI_D0);
- pci_restore_state(pci);
+ if (pci_restore_state(pci) < 0) {
+ printk(KERN_ERR "cs5535audio: pci_restore_state failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
if (pci_enable_device(pci) < 0) {
printk(KERN_ERR "cs5535audio: pci_enable_device failed, "
"disabling device\n");
@@ -112,17 +120,17 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
if (!timeout)
snd_printk(KERN_ERR "Failure getting AC Link ready\n");
- /* we depend on ac97 to perform the codec power up */
- snd_ac97_resume(cs5535au->ac97);
/* set up rate regs, dma. actual initiation is done in trig */
for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
struct cs5535audio_dma *dma = &cs5535au->dmas[i];
- if (dma && dma->substream && dma->suspended) {
+ if (dma && dma->substream) {
dma->substream->ops->prepare(dma->substream);
dma->ops->setup_prd(cs5535au, dma->saved_prd);
}
}
-
+
+ /* we depend on ac97 to perform the codec power up */
+ snd_ac97_resume(cs5535au->ac97);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index f27b6a733b96..499ee1a5319d 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1595,15 +1595,7 @@ static struct snd_kcontrol_new snd_echo_clock_source_switch __devinitdata = {
#ifdef ECHOCARD_HAS_PHANTOM_POWER
/******************* Phantom power switch *******************/
-static int snd_echo_phantom_power_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 snd_echo_phantom_power_info snd_ctl_boolean_mono_info
static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1646,15 +1638,7 @@ static struct snd_kcontrol_new snd_echo_phantom_power_switch __devinitdata = {
#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
/******************* Digital input automute switch *******************/
-static int snd_echo_automute_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 snd_echo_automute_info snd_ctl_boolean_mono_info
static int snd_echo_automute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1695,18 +1679,7 @@ static struct snd_kcontrol_new snd_echo_automute_switch __devinitdata = {
/******************* VU-meters switch *******************/
-static int snd_echo_vumeters_switch_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct echoaudio *chip;
-
- chip = snd_kcontrol_chip(kcontrol);
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_echo_vumeters_switch_info snd_ctl_boolean_mono_info
static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 42afa837d9b4..e6c100770392 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -43,11 +43,11 @@ static int wait_handshake(struct echoaudio *chip)
{
int i;
- /* Wait up to 10ms for the handshake from the DSP */
+ /* Wait up to 20ms for the handshake from the DSP */
for (i = 0; i < HANDSHAKE_TIMEOUT; i++) {
/* Look for the handshake value */
+ barrier();
if (chip->comm_page->handshake) {
- /*if (i) DE_ACT(("Handshake time: %d\n", i));*/
return 0;
}
udelay(1);
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
index e55ee00991ac..e352f3ae292c 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.h
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -642,18 +642,18 @@ struct comm_page { /* Base Length*/
u32 flags; /* See Appendix A below 0x004 4 */
u32 unused; /* Unused entry 0x008 4 */
u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */
- volatile u32 handshake; /* DSP command handshake 0x010 4 */
+ u32 handshake; /* DSP command handshake 0x010 4 */
u32 cmd_start; /* Chs. to start mask 0x014 4 */
u32 cmd_stop; /* Chs. to stop mask 0x018 4 */
u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */
u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */
struct sg_entry sglist_addr[DSP_MAXPIPES];
/* Chs. Physical sglist addrs 0x060 32*8 */
- volatile u32 position[DSP_MAXPIPES];
+ u32 position[DSP_MAXPIPES];
/* Positions for ea. ch. 0x160 32*4 */
- volatile s8 vu_meter[DSP_MAXPIPES];
+ s8 vu_meter[DSP_MAXPIPES];
/* VU meters 0x1e0 32*1 */
- volatile s8 peak_meter[DSP_MAXPIPES];
+ s8 peak_meter[DSP_MAXPIPES];
/* Peak meters 0x200 32*1 */
s8 line_out_level[DSP_MAXAUDIOOUTPUTS];
/* Output gain 0x220 16*1 */
@@ -665,7 +665,7 @@ struct comm_page { /* Base Length*/
/* Gina/Darla play filters - obsolete 0x3c0 168*4 */
u32 rec_coeff[MAX_REC_TAPS];
/* Gina/Darla record filters - obsolete 0x660 192*4 */
- volatile u16 midi_input[MIDI_IN_BUFFER_SIZE];
+ u16 midi_input[MIDI_IN_BUFFER_SIZE];
/* MIDI input data transfer buffer 0x960 256*2 */
u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */
u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */
@@ -674,11 +674,10 @@ struct comm_page { /* Base Length*/
u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */
u16 input_clock; /* Chg. Input clock state 0xb68 2 */
u16 output_clock; /* Chg. Output clock state 0xb6a 2 */
- volatile u32 status_clocks;
- /* Current Input clock state 0xb6c 4 */
+ u32 status_clocks; /* Current Input clock state 0xb6c 4 */
u32 ext_box_status; /* External box status 0xb70 4 */
u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */
- volatile u32 midi_out_free_count;
+ u32 midi_out_free_count;
/* # of bytes free in MIDI output FIFO 0xb78 4 */
u32 unused2; /* Cyclic pipes 0xb7c 4 */
u32 control_register;
diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile
index e521c38cef45..cf2d5636d8be 100644
--- a/sound/pci/emu10k1/Makefile
+++ b/sound/pci/emu10k1/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-emu10k1-objs := emu10k1.o emu10k1_main.o \
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 55caf341933a..9680caff90c8 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -1,6 +1,6 @@
/*
* The driver for the EMU10K1 (SB Live!) based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
* Added support for Audigy 2 Value.
@@ -32,7 +32,7 @@
#include <sound/emu10k1.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("EMU10K1");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS},"
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 404ae1be0a4b..97c41d72a255 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips
*
@@ -31,6 +31,8 @@
*
*/
+#include <linux/sched.h>
+#include <linux/kthread.h>
#include <sound/driver.h>
#include <linux/delay.h>
#include <linux/init.h>
@@ -702,6 +704,65 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
return 0;
}
+int emu1010_firmware_thread(void *data) {
+ struct snd_emu10k1 * emu = data;
+ int tmp,tmp2;
+ int reg;
+ int err;
+
+ for (;;) {
+ /* Delay to allow Audio Dock to settle */
+ msleep(1000);
+ if (kthread_should_stop())
+ break;
+ snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */
+ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) {
+ /* Audio Dock attached */
+ /* Return to Audio Dock programming mode */
+ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
+ if (emu->card_capabilities->emu1010 == 1) {
+ if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 2) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 3) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ }
+
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 );
+ snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
+ snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
+ /* ID, should read & 0x7f = 0x55 when FPGA programmed. */
+ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
+ if ((reg & 0x1f) != 0x15) {
+ /* FPGA failed to be programmed */
+ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
+ return 0;
+ return -ENODEV;
+ }
+ snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
+ snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp );
+ snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 );
+ snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2);
+ /* Sync clocking between 1010 and Dock */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all. Default is muted after a firmware load */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+ }
+ }
+ return 0;
+}
+
/*
* EMU-1010 - details found out from this driver, official MS Win drivers,
* testing the card:
@@ -817,8 +878,16 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg);
snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp );
- /* ADAT input. */
- snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x01 );
+ /* Optical -> ADAT I/O */
+ /* 0 : SPDIF
+ * 1 : ADAT
+ */
+ emu->emu1010.optical_in = 1; /* IN_ADAT */
+ emu->emu1010.optical_out = 1; /* IN_ADAT */
+ tmp = 0;
+ tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) |
+ (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0);
+ snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp );
snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp );
/* Set no attenuation on Audio Dock pads. */
snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 );
@@ -1004,49 +1073,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp );
snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */
- /* Delay to allow Audio Dock to settle */
- msleep(100);
- snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */
- snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */
- /* FIXME: The loading of this should be able to happen any time,
- * as the user can plug/unplug it at any time
- */
- if (reg & (EMU_HANA_OPTION_DOCK_ONLINE | EMU_HANA_OPTION_DOCK_OFFLINE) ) {
- /* Audio Dock attached */
- /* Return to Audio Dock programming mode */
- snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
- if (emu->card_capabilities->emu1010 == 1) {
- if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
- return err;
- }
- } else if (emu->card_capabilities->emu1010 == 2) {
- if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
- return err;
- }
- } else if (emu->card_capabilities->emu1010 == 3) {
- if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
- return err;
- }
- }
+ /* Start Micro/Audio Dock firmware loader thread */
+ emu->emu1010.firmware_thread = kthread_create(&emu1010_firmware_thread,
+ emu,
+ "emu1010_firmware");
+ wake_up_process(emu->emu1010.firmware_thread);
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 );
- snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
- snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
- /* ID, should read & 0x7f = 0x55 when FPGA programmed. */
- snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
- snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
- if ((reg & 0x3f) != 0x15) {
- /* FPGA failed to be programmed */
- snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
- return 0;
- return -ENODEV;
- }
- snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
- snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp );
- snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 );
- snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2);
- }
#if 0
snd_emu1010_fpga_link_dst_src_write(emu,
EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32B + 2); /* ALICE2 bus 0xa2 */
@@ -1132,7 +1164,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
emu->emu1010.output_source[23] = 28;
/* TEMP: Select SPDIF in/out */
- snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */
+ //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */
/* TEMP: Select 48kHz SPDIF out */
snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */
@@ -1173,6 +1205,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
if (emu->card_capabilities->emu1010) {
/* Disable 48Volt power to Audio Dock */
snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 );
+ kthread_stop(emu->emu1010.firmware_thread);
}
if (emu->memhdr)
snd_util_memhdr_free(emu->memhdr);
@@ -1722,8 +1755,9 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
goto error;
}
- emu->page_ptr_table = (void **)vmalloc(emu->max_cache_pages * sizeof(void*));
- emu->page_addr_table = (unsigned long*)vmalloc(emu->max_cache_pages * sizeof(unsigned long));
+ emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *));
+ emu->page_addr_table = vmalloc(emu->max_cache_pages *
+ sizeof(unsigned long));
if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) {
err = -ENOMEM;
goto error;
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index e4af7a9b808c..1ec7ebaff9e9 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1062,14 +1062,7 @@ static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu)
return 0;
}
-static int snd_emu10k1x_shared_spdif_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 snd_emu10k1x_shared_spdif_info snd_ctl_boolean_mono_info
static int snd_emu10k1x_shared_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 7206c0fa06f2..9bf1cd592199 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for effect processor FX8010
*
@@ -642,10 +642,8 @@ snd_emu10k1_look_for_ctl(struct snd_emu10k1 *emu, struct snd_ctl_elem_id *id)
{
struct snd_emu10k1_fx8010_ctl *ctl;
struct snd_kcontrol *kcontrol;
- struct list_head *list;
-
- list_for_each(list, &emu->fx8010.gpr_ctl) {
- ctl = emu10k1_gpr_ctl(list);
+
+ list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) {
kcontrol = ctl->kcontrol;
if (kcontrol->id.iface == id->iface &&
!strcmp(kcontrol->id.name, id->name) &&
@@ -895,14 +893,12 @@ static int snd_emu10k1_list_controls(struct snd_emu10k1 *emu,
struct snd_emu10k1_fx8010_control_gpr *gctl;
struct snd_emu10k1_fx8010_ctl *ctl;
struct snd_ctl_elem_id *id;
- struct list_head *list;
gctl = kmalloc(sizeof(*gctl), GFP_KERNEL);
if (! gctl)
return -ENOMEM;
- list_for_each(list, &emu->fx8010.gpr_ctl) {
- ctl = emu10k1_gpr_ctl(list);
+ list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) {
total++;
if (icode->gpr_list_controls &&
i < icode->gpr_list_control_count) {
@@ -1207,7 +1203,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu)
A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT));
snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100);
gpr += 2;
-
+
/* PCM Surround Playback (independent from stereo mix) */
A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_REAR));
A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_REAR));
@@ -1267,8 +1263,16 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
/* emu1212 DSP 0 and DSP 1 Capture */
if (emu->card_capabilities->emu1010) {
- A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0));
- A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1));
+ if (emu->card_capabilities->ca0108_chip) {
+ /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */
+ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001);
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_GPR(tmp));
+ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x1), A_C_00000001);
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr), A_GPR(tmp));
+ } else {
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0));
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1));
+ }
snd_emu10k1_init_stereo_control(&controls[nctl++], "EMU Capture Volume", gpr, 0);
gpr += 2;
}
@@ -1516,7 +1520,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
/* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */
snd_printk("EMU outputs on\n");
for (z = 0; z < 8; z++) {
- A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ if (emu->card_capabilities->ca0108_chip) {
+ A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ } else {
+ A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ }
}
}
@@ -1557,106 +1565,116 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
#endif
if (emu->card_capabilities->emu1010) {
- snd_printk("EMU inputs on\n");
- /* Capture 16 (originally 8) channels of S32_LE sound */
-
- /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
- /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
- /* A_P16VIN(0) is delayed by one sample,
- * so all other A_P16VIN channels will need to also be delayed
- */
- /* Left ADC in. 1 of 2 */
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
- /* Right ADC in 1 of 2 */
- gpr_map[gpr++] = 0x00000000;
- /* Delaying by one sample: instead of copying the input
- * value A_P16VIN to output A_FXBUS2 as in the first channel,
- * we use an auxiliary register, delaying the value by one
- * sample
- */
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000);
- /* For 96kHz mode */
- /* Left ADC in. 2 of 2 */
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000);
- /* Right ADC in 2 of 2 */
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
- /* Pavel Hofman - we still have voices, A_FXBUS2s, and
- * A_P16VINs available -
- * let's add 8 more capture channels - total of 16
- */
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x10));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x12));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x14));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x16));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x18));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x1a));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x1c));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x1e));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
- A_C_00000000, A_C_00000000);
+ if (emu->card_capabilities->ca0108_chip) {
+ snd_printk("EMU2 inputs on\n");
+ for (z = 0; z < 0x10; z++) {
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp,
+ bit_shifter16,
+ A3_EMU32IN(z),
+ A_FXBUS2(z*2) );
+ }
+ } else {
+ snd_printk("EMU inputs on\n");
+ /* Capture 16 (originally 8) channels of S32_LE sound */
+
+ /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
+ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
+ /* A_P16VIN(0) is delayed by one sample,
+ * so all other A_P16VIN channels will need to also be delayed
+ */
+ /* Left ADC in. 1 of 2 */
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
+ /* Right ADC in 1 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ /* Delaying by one sample: instead of copying the input
+ * value A_P16VIN to output A_FXBUS2 as in the first channel,
+ * we use an auxiliary register, delaying the value by one
+ * sample
+ */
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000);
+ /* For 96kHz mode */
+ /* Left ADC in. 2 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000);
+ /* Right ADC in 2 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
+ /* Pavel Hofman - we still have voices, A_FXBUS2s, and
+ * A_P16VINs available -
+ * let's add 8 more capture channels - total of 16
+ */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x10));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x12));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x14));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x16));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x18));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1a));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1c));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1e));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
+ A_C_00000000, A_C_00000000);
+ }
#if 0
for (z = 4; z < 8; z++) {
@@ -2418,14 +2436,13 @@ static void copy_string(char *dst, char *src, char *null, int idx)
strcpy(dst, src);
}
-static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
+static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
struct snd_emu10k1_fx8010_info *info)
{
char **fxbus, **extin, **extout;
unsigned short fxbus_mask, extin_mask, extout_mask;
int res;
- memset(info, 0, sizeof(info));
info->internal_tram_size = emu->fx8010.itram_size;
info->external_tram_size = emu->fx8010.etram_pages.bytes / 2;
fxbus = fxbuses;
@@ -2442,7 +2459,6 @@ static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
for (res = 16; res < 32; res++, extout++)
copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res);
info->gpr_controls = emu->fx8010.gpr_count;
- return 0;
}
static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg)
@@ -2463,10 +2479,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un
info = kmalloc(sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
- if ((res = snd_emu10k1_fx8010_info(emu, info)) < 0) {
- kfree(info);
- return res;
- }
+ snd_emu10k1_fx8010_info(emu, info);
if (copy_to_user(argp, info, sizeof(*info))) {
kfree(info);
return -EFAULT;
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 7b2c1dcc5337..54a2034d8edd 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -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>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips / mixer routines
@@ -400,15 +400,7 @@ static struct snd_kcontrol_new snd_emu1010_input_enum_ctls[] __devinitdata = {
-
-static int snd_emu1010_adc_pads_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 snd_emu1010_adc_pads_info snd_ctl_boolean_mono_info
static int snd_emu1010_adc_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -456,14 +448,7 @@ static struct snd_kcontrol_new snd_emu1010_adc_pads[] __devinitdata = {
EMU1010_ADC_PADS("ADC1 14dB PAD 0202 Capture Switch", EMU_HANA_0202_ADC_PAD1),
};
-static int snd_emu1010_dac_pads_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 snd_emu1010_dac_pads_info snd_ctl_boolean_mono_info
static int snd_emu1010_dac_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -516,17 +501,19 @@ static struct snd_kcontrol_new snd_emu1010_dac_pads[] __devinitdata = {
static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[2] = {
- "44100", "48000"
+ static char *texts[4] = {
+ "44100", "48000", "SPDIF", "ADAT"
};
-
+
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item > 1)
- uinfo->value.enumerated.item = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
+
+
}
static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol,
@@ -584,6 +571,44 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol,
/* Unmute all */
snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
break;
+
+ case 2: /* Take clock from S/PDIF IN */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, sync to S/PDIF input */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2,
+ EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+
+ case 3:
+ /* Take clock from ADAT IN */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, sync to ADAT input */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+
+
+ break;
}
}
return change;
@@ -871,7 +896,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control =
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
- .count = 4,
+ .count = 3,
.info = snd_emu10k1_spdif_info,
.get = snd_emu10k1_spdif_get_mask
};
@@ -880,7 +905,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_control =
{
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
- .count = 4,
+ .count = 3,
.info = snd_emu10k1_spdif_info,
.get = snd_emu10k1_spdif_get,
.put = snd_emu10k1_spdif_put
@@ -1326,14 +1351,7 @@ static struct snd_kcontrol_new snd_emu10k1_efx_attn_control =
.put = snd_emu10k1_efx_attn_put
};
-static int snd_emu10k1_shared_spdif_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 snd_emu10k1_shared_spdif_info snd_ctl_boolean_mono_info
static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c
index 950c6bcd6b7d..04c7cf703531 100644
--- a/sound/pci/emu10k1/emumpu401.c
+++ b/sound/pci/emu10k1/emumpu401.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of EMU10K1 MPU-401 in UART mode
*
*
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index eda5cb373ded..5ce5befc701b 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips / PCM routines
* Multichannel PCM support Copyright (c) Lee Revell <rlrevell@joe-job.com>
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index 2c1585991bc8..c3fb10e81c9e 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips / proc interface routines
*
@@ -240,8 +240,42 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_emu10k1 *emu = entry->private_data;
- snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS);
- snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS);
+ u32 value;
+ u32 value2;
+ unsigned long flags;
+ u32 rate;
+
+ if (emu->card_capabilities->emu1010) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x38, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ if ((value & 0x1) == 0) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x2a, &value);
+ snd_emu1010_fpga_read(emu, 0x2b, &value2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ rate = 0x1770000 / (((value << 5) | value2)+1);
+ snd_iprintf(buffer, "ADAT Locked : %u\n", rate);
+ } else {
+ snd_iprintf(buffer, "ADAT Unlocked\n");
+ }
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x20, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ if ((value & 0x4) == 0) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x28, &value);
+ snd_emu1010_fpga_read(emu, 0x29, &value2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ rate = 0x1770000 / (((value << 5) | value2)+1);
+ snd_iprintf(buffer, "SPDIF Locked : %d\n", rate);
+ } else {
+ snd_iprintf(buffer, "SPDIF Unlocked\n");
+ }
+ } else {
+ snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS);
+ snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS);
+ }
#if 0
val = snd_emu10k1_ptr_read(emu, ZVSRCS, 0);
snd_iprintf(buffer, "\nZoomed Video\n");
@@ -379,20 +413,16 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_emu10k1 *emu = entry->private_data;
- unsigned long value;
+ int value;
unsigned long flags;
- unsigned long regs;
int i;
snd_iprintf(buffer, "EMU1010 Registers:\n\n");
- for(i = 0; i < 0x30; i+=1) {
+ for(i = 0; i < 0x40; i+=1) {
spin_lock_irqsave(&emu->emu_lock, flags);
- regs=i+0x40; /* 0x40 upwards are registers. */
- outl(regs, emu->port + A_IOCFG);
- outl(regs | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */
- value = inl(emu->port + A_IOCFG);
+ snd_emu1010_fpga_read(emu, i, &value);
spin_unlock_irqrestore(&emu->emu_lock, flags);
- snd_iprintf(buffer, "%02X: %08lX, %02lX\n", i, value, (value >> 8) & 0x7f);
+ snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f);
}
}
@@ -555,9 +585,9 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu)
{
struct snd_info_entry *entry;
#ifdef CONFIG_SND_DEBUG
- if ((emu->card_capabilities->emu1010) &&
- snd_card_proc_new(emu->card, "emu1010_regs", &entry)) {
- snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read);
+ if (emu->card_capabilities->emu1010) {
+ if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read);
}
if (! snd_card_proc_new(emu->card, "io_regs", &entry)) {
snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read);
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index 116e1c8d9361..6702c15fefa3 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips
*
@@ -226,9 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu,
return 0;
}
-int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value)
+int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value)
{
- if (reg < 0 || reg > 0x3f)
+ if (reg > 0x3f)
return 1;
reg += 0x40; /* 0x40 upwards are registers. */
if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */
@@ -244,9 +244,9 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value)
return 0;
}
-int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value)
+int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value)
{
- if (reg < 0 || reg > 0x3f)
+ if (reg > 0x3f)
return 1;
reg += 0x40; /* 0x40 upwards are registers. */
outl(reg, emu->port + A_IOCFG);
@@ -261,7 +261,7 @@ int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value)
/* Each Destination has one and only one Source,
* but one Source can feed any number of Destinations simultaneously.
*/
-int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src)
+int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src)
{
snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) );
snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) );
diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c
index 4f18f7e8bcfb..3c114b45e0b2 100644
--- a/sound/pci/emu10k1/irq.c
+++ b/sound/pci/emu10k1/irq.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for IRQ control of EMU10K1 chips
*
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 4fcaefe5a3c5..48097c6bb15c 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Copyright (c) by Takashi Iwai <tiwai@suse.de>
*
* EMU10K1 memory page allocation (PTB area)
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 7ee19c63c2c8..d619a3842cdd 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -124,11 +124,12 @@
/* hardware definition */
static struct snd_pcm_hardware snd_p16v_playback_hw = {
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_MMAP_VALID),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
.formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */
.rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
.rate_min = 44100,
@@ -207,6 +208,11 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
+ runtime->sync.id32[0] = substream->pcm->card->number;
+ runtime->sync.id32[1] = 'P';
+ runtime->sync.id32[2] = 16;
+ runtime->sync.id32[3] = 'V';
+
return 0;
}
/* open_capture callback */
@@ -448,6 +454,9 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream,
break;
}
snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
runtime = s->runtime;
epcm = runtime->private_data;
channel = substream->pcm->device-emu->p16v_device_offset;
diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c
index 1db50fe61475..04fa8492abb0 100644
--- a/sound/pci/emu10k1/voice.c
+++ b/sound/pci/emu10k1/voice.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Lee Revell <rlrevell@joe-job.com>
* Routines for control of EMU10K1 chips - voice manager
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 21cb4268a59b..b958f869cb13 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1,6 +1,6 @@
/*
* Driver for Ensoniq ES1370/ES1371 AudioPCI soundcard
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Thomas Sailer <sailer@ife.ee.ethz.ch>
*
* This program is free software; you can redistribute it and/or modify
@@ -61,7 +61,7 @@
#endif
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>");
MODULE_LICENSE("GPL");
#ifdef CHIP1370
MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370");
@@ -1419,15 +1419,7 @@ static int snd_ens1373_spdif_stream_put(struct snd_kcontrol *kcontrol,
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_es1371_spdif_info, \
.get = snd_es1371_spdif_get, .put = snd_es1371_spdif_put }
-static int snd_es1371_spdif_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 snd_es1371_spdif_info snd_ctl_boolean_mono_info
static int snd_es1371_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1489,15 +1481,7 @@ static struct snd_kcontrol_new snd_es1371_mixer_spdif[] __devinitdata = {
};
-static int snd_es1373_rear_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 snd_es1373_rear_info snd_ctl_boolean_mono_info
static int snd_es1373_rear_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1542,15 +1526,7 @@ static struct snd_kcontrol_new snd_ens1373_rear __devinitdata =
.put = snd_es1373_rear_put,
};
-static int snd_es1373_line_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 snd_es1373_line_info snd_ctl_boolean_mono_info
static int snd_es1373_line_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1707,15 +1683,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq,
.get = snd_ensoniq_control_get, .put = snd_ensoniq_control_put, \
.private_value = mask }
-static int snd_ensoniq_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 snd_ensoniq_control_info snd_ctl_boolean_mono_info
static int snd_ensoniq_control_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index fec29a108945..fb25abe68a02 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1,7 +1,7 @@
/*
* Driver for ESS Solo-1 (ES1938, ES1946, ES1969) soundcard
* Copyright (c) by Jaromir Koutek <miri@punknet.cz>,
- * Jaroslav Kysela <perex@suse.cz>,
+ * Jaroslav Kysela <perex@perex.cz>,
* Thomas Sailer <sailer@ife.ee.ethz.ch>,
* Abramo Bagnara <abramo@alsa-project.org>,
* Markus Gruber <gruber@eikon.tum.de>
@@ -1066,15 +1066,7 @@ static int snd_es1938_put_mux(struct snd_kcontrol *kcontrol,
return snd_es1938_mixer_bits(chip, 0x1c, 0x07, val) != val;
}
-static int snd_es1938_info_spatializer_enable(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 snd_es1938_info_spatializer_enable snd_ctl_boolean_mono_info
static int snd_es1938_get_spatializer_enable(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1120,15 +1112,7 @@ static int snd_es1938_get_hw_volume(struct snd_kcontrol *kcontrol,
return 0;
}
-static int snd_es1938_info_hw_switch(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 snd_es1938_info_hw_switch snd_ctl_boolean_stereo_info
static int snd_es1938_get_hw_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 2faf009076bb..d69b11d1f993 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -843,10 +843,9 @@ static void snd_es1968_bob_dec(struct es1968 *chip)
snd_es1968_bob_stop(chip);
else if (chip->bob_freq > ESM_BOB_FREQ) {
/* check reduction of timer frequency */
- struct list_head *p;
int max_freq = ESM_BOB_FREQ;
- list_for_each(p, &chip->substream_list) {
- struct esschan *es = list_entry(p, struct esschan, list);
+ struct esschan *es;
+ list_for_each_entry(es, &chip->substream_list, list) {
if (max_freq < es->bob_freq)
max_freq = es->bob_freq;
}
@@ -1316,12 +1315,11 @@ static struct snd_pcm_hardware snd_es1968_capture = {
static int calc_available_memory_size(struct es1968 *chip)
{
- struct list_head *p;
int max_size = 0;
-
+ struct esm_memory *buf;
+
mutex_lock(&chip->memory_mutex);
- list_for_each(p, &chip->buf_list) {
- struct esm_memory *buf = list_entry(p, struct esm_memory, list);
+ list_for_each_entry(buf, &chip->buf_list, list) {
if (buf->empty && buf->buf.bytes > max_size)
max_size = buf->buf.bytes;
}
@@ -1335,12 +1333,10 @@ static int calc_available_memory_size(struct es1968 *chip)
static struct esm_memory *snd_es1968_new_memory(struct es1968 *chip, int size)
{
struct esm_memory *buf;
- struct list_head *p;
-
+
size = ALIGN(size, ESM_MEM_ALIGN);
mutex_lock(&chip->memory_mutex);
- list_for_each(p, &chip->buf_list) {
- buf = list_entry(p, struct esm_memory, list);
+ list_for_each_entry(buf, &chip->buf_list, list) {
if (buf->empty && buf->buf.bytes >= size)
goto __found;
}
@@ -1938,10 +1934,9 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
}
if (event & ESM_SOUND_IRQ) {
- struct list_head *p;
+ struct esschan *es;
spin_lock(&chip->substream_lock);
- list_for_each(p, &chip->substream_list) {
- struct esschan *es = list_entry(p, struct esschan, list);
+ list_for_each_entry(es, &chip->substream_list, list) {
if (es->running)
snd_es1968_update_pcm(chip, es);
}
@@ -2345,7 +2340,7 @@ static int es1968_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct es1968 *chip = card->private_data;
- struct list_head *p;
+ struct esschan *es;
if (! chip->do_pm)
return 0;
@@ -2374,8 +2369,7 @@ static int es1968_resume(struct pci_dev *pci)
/* restore ac97 state */
snd_ac97_resume(chip->ac97);
- list_for_each(p, &chip->substream_list) {
- struct esschan *es = list_entry(p, struct esschan, list);
+ list_for_each_entry(es, &chip->substream_list, list) {
switch (es->mode) {
case ESM_MODE_PLAY:
snd_es1968_playback_setup(chip, es, es->substream->runtime);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 11015178e207..9939109f05a2 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1,6 +1,6 @@
/*
* The driver for the ForteMedia FM801 based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* Support FM only card by Andy Shevchenko <andy@smile.org.ua>
*
@@ -42,7 +42,7 @@
#define TEA575X_RADIO 1
#endif
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("ForteMedia FM801");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801},"
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index b2484bbdcc1d..ab0c726d648e 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,19 +1,18 @@
-snd-hda-intel-objs := hda_intel.o
+snd-hda-intel-y := hda_intel.o
# since snd-hda-intel is the only driver using hda-codec,
# merge it into a single module although it was originally
# designed to be individual modules
-snd-hda-intel-objs += hda_codec.o \
- hda_generic.o \
- patch_realtek.o \
- patch_cmedia.o \
- patch_analog.o \
- patch_sigmatel.o \
- patch_si3054.o \
- patch_atihdmi.o \
- patch_conexant.o \
- patch_via.o
-ifdef CONFIG_PROC_FS
-snd-hda-intel-objs += hda_proc.o
-endif
+snd-hda-intel-y += hda_codec.o
+snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
+snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
+snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index f87f8f088956..187533e477c6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -31,7 +31,15 @@
#include <sound/tlv.h>
#include <sound/initval.h>
#include "hda_local.h"
-
+#include <sound/hda_hwdep.h>
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* define this option here to hide as static */
+static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+ "(in second, 0 = disable).");
+#endif
/*
* vendor / preset table
@@ -59,6 +67,13 @@ static struct hda_vendor_id hda_vendor_ids[] = {
#include "hda_patch.h"
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_power_work(struct work_struct *work);
+static void hda_keep_power_on(struct hda_codec *codec);
+#else
+static inline void hda_keep_power_on(struct hda_codec *codec) {}
+#endif
+
/**
* snd_hda_codec_read - send a command and get the response
* @codec: the HDA codec
@@ -76,12 +91,14 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
unsigned int verb, unsigned int parm)
{
unsigned int res;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
if (!codec->bus->ops.command(codec, nid, direct, verb, parm))
res = codec->bus->ops.get_response(codec);
else
res = (unsigned int)-1;
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return res;
}
@@ -101,9 +118,11 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
unsigned int verb, unsigned int parm)
{
int err;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
err = codec->bus->ops.command(codec, nid, direct, verb, parm);
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return err;
}
@@ -136,6 +155,8 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
unsigned int parm;
parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT);
+ if (parm == -1)
+ return 0;
*start_id = (parm >> 16) & 0x7fff;
return (int)(parm & 0x7fff);
}
@@ -387,6 +408,13 @@ int __devinit snd_hda_bus_new(struct snd_card *card,
return 0;
}
+#ifdef CONFIG_SND_HDA_GENERIC
+#define is_generic_config(codec) \
+ (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic"))
+#else
+#define is_generic_config(codec) 0
+#endif
+
/*
* find a matching codec preset
*/
@@ -395,7 +423,7 @@ find_codec_preset(struct hda_codec *codec)
{
const struct hda_codec_preset **tbl, *preset;
- if (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic"))
+ if (is_generic_config(codec))
return NULL; /* use the generic parser */
for (tbl = hda_preset_tables; *tbl; tbl++) {
@@ -486,6 +514,10 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node)
}
+static void init_hda_cache(struct hda_cache_rec *cache,
+ unsigned int record_size);
+static void free_hda_cache(struct hda_cache_rec *cache);
+
/*
* codec destructor
*/
@@ -493,17 +525,20 @@ static void snd_hda_codec_free(struct hda_codec *codec)
{
if (!codec)
return;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+ flush_scheduled_work();
+#endif
list_del(&codec->list);
codec->bus->caddr_tbl[codec->addr] = NULL;
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
- kfree(codec->amp_info);
+ free_hda_cache(&codec->amp_cache);
+ free_hda_cache(&codec->cmd_cache);
kfree(codec->wcaps);
kfree(codec);
}
-static void init_amp_hash(struct hda_codec *codec);
-
/**
* snd_hda_codec_new - create a HDA codec
* @bus: the bus to assign
@@ -537,7 +572,17 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
codec->bus = bus;
codec->addr = codec_addr;
mutex_init(&codec->spdif_mutex);
- init_amp_hash(codec);
+ init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
+ init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
+ /* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
+ * the caller has to power down appropriatley after initialization
+ * phase.
+ */
+ hda_keep_power_on(codec);
+#endif
list_add_tail(&codec->list, &bus->codec_list);
bus->caddr_tbl[codec_addr] = codec;
@@ -581,10 +626,26 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
snd_hda_get_codec_name(codec, bus->card->mixername,
sizeof(bus->card->mixername));
- if (codec->preset && codec->preset->patch)
- err = codec->preset->patch(codec);
- else
+#ifdef CONFIG_SND_HDA_GENERIC
+ if (is_generic_config(codec)) {
err = snd_hda_parse_generic_codec(codec);
+ goto patched;
+ }
+#endif
+ if (codec->preset && codec->preset->patch) {
+ err = codec->preset->patch(codec);
+ goto patched;
+ }
+
+ /* call the default parser */
+#ifdef CONFIG_SND_HDA_GENERIC
+ err = snd_hda_parse_generic_codec(codec);
+#else
+ printk(KERN_ERR "hda-codec: No codec parser is available\n");
+ err = -ENODEV;
+#endif
+
+ patched:
if (err < 0) {
snd_hda_codec_free(codec);
return err;
@@ -594,6 +655,9 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
init_unsol_queue(bus);
snd_hda_codec_proc_new(codec);
+#ifdef CONFIG_SND_HDA_HWDEP
+ snd_hda_create_hwdep(codec);
+#endif
sprintf(component, "HDA:%08x", codec->vendor_id);
snd_component_add(codec->bus->card, component);
@@ -637,59 +701,72 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
#define INFO_AMP_VOL(ch) (1 << (1 + (ch)))
/* initialize the hash table */
-static void __devinit init_amp_hash(struct hda_codec *codec)
+static void __devinit init_hda_cache(struct hda_cache_rec *cache,
+ unsigned int record_size)
+{
+ memset(cache, 0, sizeof(*cache));
+ memset(cache->hash, 0xff, sizeof(cache->hash));
+ cache->record_size = record_size;
+}
+
+static void free_hda_cache(struct hda_cache_rec *cache)
{
- memset(codec->amp_hash, 0xff, sizeof(codec->amp_hash));
- codec->num_amp_entries = 0;
- codec->amp_info_size = 0;
- codec->amp_info = NULL;
+ kfree(cache->buffer);
}
/* query the hash. allocate an entry if not found. */
-static struct hda_amp_info *get_alloc_amp_hash(struct hda_codec *codec, u32 key)
+static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache,
+ u32 key)
{
- u16 idx = key % (u16)ARRAY_SIZE(codec->amp_hash);
- u16 cur = codec->amp_hash[idx];
- struct hda_amp_info *info;
+ u16 idx = key % (u16)ARRAY_SIZE(cache->hash);
+ u16 cur = cache->hash[idx];
+ struct hda_cache_head *info;
while (cur != 0xffff) {
- info = &codec->amp_info[cur];
+ info = (struct hda_cache_head *)(cache->buffer +
+ cur * cache->record_size);
if (info->key == key)
return info;
cur = info->next;
}
/* add a new hash entry */
- if (codec->num_amp_entries >= codec->amp_info_size) {
+ if (cache->num_entries >= cache->size) {
/* reallocate the array */
- int new_size = codec->amp_info_size + 64;
- struct hda_amp_info *new_info;
- new_info = kcalloc(new_size, sizeof(struct hda_amp_info),
- GFP_KERNEL);
- if (!new_info) {
+ unsigned int new_size = cache->size + 64;
+ void *new_buffer;
+ new_buffer = kcalloc(new_size, cache->record_size, GFP_KERNEL);
+ if (!new_buffer) {
snd_printk(KERN_ERR "hda_codec: "
"can't malloc amp_info\n");
return NULL;
}
- if (codec->amp_info) {
- memcpy(new_info, codec->amp_info,
- codec->amp_info_size *
- sizeof(struct hda_amp_info));
- kfree(codec->amp_info);
+ if (cache->buffer) {
+ memcpy(new_buffer, cache->buffer,
+ cache->size * cache->record_size);
+ kfree(cache->buffer);
}
- codec->amp_info_size = new_size;
- codec->amp_info = new_info;
+ cache->size = new_size;
+ cache->buffer = new_buffer;
}
- cur = codec->num_amp_entries++;
- info = &codec->amp_info[cur];
+ cur = cache->num_entries++;
+ info = (struct hda_cache_head *)(cache->buffer +
+ cur * cache->record_size);
info->key = key;
- info->status = 0; /* not initialized yet */
- info->next = codec->amp_hash[idx];
- codec->amp_hash[idx] = cur;
+ info->val = 0;
+ info->next = cache->hash[idx];
+ cache->hash[idx] = cur;
return info;
}
+/* query and allocate an amp hash entry */
+static inline struct hda_amp_info *
+get_alloc_amp_hash(struct hda_codec *codec, u32 key)
+{
+ return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key);
+}
+
/*
* query AMP capabilities for the given widget and direction
*/
@@ -700,7 +777,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
if (!info)
return 0;
- if (!(info->status & INFO_AMP_CAPS)) {
+ if (!(info->head.val & INFO_AMP_CAPS)) {
if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
nid = codec->afg;
info->amp_caps = snd_hda_param_read(codec, nid,
@@ -708,7 +785,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
AC_PAR_AMP_OUT_CAP :
AC_PAR_AMP_IN_CAP);
if (info->amp_caps)
- info->status |= INFO_AMP_CAPS;
+ info->head.val |= INFO_AMP_CAPS;
}
return info->amp_caps;
}
@@ -722,7 +799,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
if (!info)
return -EINVAL;
info->amp_caps = caps;
- info->status |= INFO_AMP_CAPS;
+ info->head.val |= INFO_AMP_CAPS;
return 0;
}
@@ -736,7 +813,7 @@ static unsigned int get_vol_mute(struct hda_codec *codec,
{
u32 val, parm;
- if (info->status & INFO_AMP_VOL(ch))
+ if (info->head.val & INFO_AMP_VOL(ch))
return info->vol[ch];
parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
@@ -745,7 +822,7 @@ static unsigned int get_vol_mute(struct hda_codec *codec,
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val & 0xff;
- info->status |= INFO_AMP_VOL(ch);
+ info->head.val |= INFO_AMP_VOL(ch);
return info->vol[ch];
}
@@ -792,12 +869,50 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
return 0;
val &= mask;
val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
- if (info->vol[ch] == val && !codec->in_resume)
+ if (info->vol[ch] == val)
return 0;
put_vol_mute(codec, info, nid, ch, direction, idx, val);
return 1;
}
+/*
+ * update the AMP stereo with the same mask and value
+ */
+int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
+ int direction, int idx, int mask, int val)
+{
+ int ch, ret = 0;
+ for (ch = 0; ch < 2; ch++)
+ ret |= snd_hda_codec_amp_update(codec, nid, ch, direction,
+ idx, mask, val);
+ return ret;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/* resume the all amp commands from the cache */
+void snd_hda_codec_resume_amp(struct hda_codec *codec)
+{
+ struct hda_amp_info *buffer = codec->amp_cache.buffer;
+ int i;
+
+ for (i = 0; i < codec->amp_cache.size; i++, buffer++) {
+ u32 key = buffer->head.key;
+ hda_nid_t nid;
+ unsigned int idx, dir, ch;
+ if (!key)
+ continue;
+ nid = key & 0xff;
+ idx = (key >> 16) & 0xff;
+ dir = (key >> 24) & 0xff;
+ for (ch = 0; ch < 2; ch++) {
+ if (!(buffer->head.val & INFO_AMP_VOL(ch)))
+ continue;
+ put_vol_mute(codec, buffer, nid, ch, dir, idx,
+ buffer->vol[ch]);
+ }
+ }
+}
+#endif /* SND_HDA_NEEDS_RESUME */
/*
* AMP control callbacks
@@ -844,9 +959,11 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
if (chs & 1)
- *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f;
+ *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx)
+ & HDA_AMP_VOLMASK;
if (chs & 2)
- *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f;
+ *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx)
+ & HDA_AMP_VOLMASK;
return 0;
}
@@ -861,6 +978,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change = 0;
+ snd_hda_power_up(codec);
if (chs & 1) {
change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
0x7f, *valp);
@@ -869,6 +987,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
if (chs & 2)
change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
0x7f, *valp);
+ snd_hda_power_down(codec);
return change;
}
@@ -923,10 +1042,10 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
if (chs & 1)
*valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) &
- 0x80) ? 0 : 1;
+ HDA_AMP_MUTE) ? 0 : 1;
if (chs & 2)
*valp = (snd_hda_codec_amp_read(codec, nid, 1, dir, idx) &
- 0x80) ? 0 : 1;
+ HDA_AMP_MUTE) ? 0 : 1;
return 0;
}
@@ -941,15 +1060,22 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change = 0;
+ snd_hda_power_up(codec);
if (chs & 1) {
change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
- 0x80, *valp ? 0 : 0x80);
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
valp++;
}
if (chs & 2)
change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
- 0x80, *valp ? 0 : 0x80);
-
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (codec->patch_ops.check_power_status)
+ codec->patch_ops.check_power_status(codec, nid);
+#endif
+ snd_hda_power_down(codec);
return change;
}
@@ -1002,6 +1128,93 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
}
/*
+ * generic bound volume/swtich controls
+ */
+int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ kcontrol->private_value = *c->values;
+ err = c->ops->info(kcontrol, uinfo);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ kcontrol->private_value = *c->values;
+ err = c->ops->get(kcontrol, ucontrol);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ unsigned long *vals;
+ int err = 0, change = 0;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ for (vals = c->values; *vals; vals++) {
+ kcontrol->private_value = *vals;
+ err = c->ops->put(kcontrol, ucontrol);
+ if (err < 0)
+ break;
+ change |= err;
+ }
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err < 0 ? err : change;
+}
+
+int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ kcontrol->private_value = *c->values;
+ err = c->ops->tlv(kcontrol, op_flag, size, tlv);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+struct hda_ctl_ops snd_hda_bind_vol = {
+ .info = snd_hda_mixer_amp_volume_info,
+ .get = snd_hda_mixer_amp_volume_get,
+ .put = snd_hda_mixer_amp_volume_put,
+ .tlv = snd_hda_mixer_amp_tlv
+};
+
+struct hda_ctl_ops snd_hda_bind_sw = {
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = snd_hda_mixer_amp_switch_put,
+ .tlv = snd_hda_mixer_amp_tlv
+};
+
+/*
* SPDIF out controls
*/
@@ -1118,26 +1331,20 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
change = codec->spdif_ctls != val;
codec->spdif_ctls = val;
- if (change || codec->in_resume) {
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- val & 0xff);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2,
- val >> 8);
+ if (change) {
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ val & 0xff);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_2,
+ val >> 8);
}
mutex_unlock(&codec->spdif_mutex);
return change;
}
-static int snd_hda_spdif_out_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 snd_hda_spdif_out_switch_info snd_ctl_boolean_mono_info
static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1161,17 +1368,16 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
change = codec->spdif_ctls != val;
- if (change || codec->in_resume) {
+ if (change) {
codec->spdif_ctls = val;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- val & 0xff);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ val & 0xff);
/* unmute amp switch (if any) */
if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
(val & AC_DIG1_ENABLE))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT |
- AC_AMP_SET_OUTPUT);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
}
mutex_unlock(&codec->spdif_mutex);
return change;
@@ -1219,8 +1425,7 @@ static struct snd_kcontrol_new dig_mixes[] = {
*
* Returns 0 if successful, or a negative error code.
*/
-int __devinit snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
- hda_nid_t nid)
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
{
int err;
struct snd_kcontrol *kctl;
@@ -1264,10 +1469,10 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol,
mutex_lock(&codec->spdif_mutex);
change = codec->spdif_in_enable != val;
- if (change || codec->in_resume) {
+ if (change) {
codec->spdif_in_enable = val;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- val);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1, val);
}
mutex_unlock(&codec->spdif_mutex);
return change;
@@ -1318,8 +1523,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = {
*
* Returns 0 if successful, or a negative error code.
*/
-int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec,
- hda_nid_t nid)
+int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
{
int err;
struct snd_kcontrol *kctl;
@@ -1338,6 +1542,79 @@ int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec,
return 0;
}
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * command cache
+ */
+
+/* build a 32bit cache key with the widget id and the command parameter */
+#define build_cmd_cache_key(nid, verb) ((verb << 8) | nid)
+#define get_cmd_cache_nid(key) ((key) & 0xff)
+#define get_cmd_cache_cmd(key) (((key) >> 8) & 0xffff)
+
+/**
+ * snd_hda_codec_write_cache - send a single command with caching
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
+ int direct, unsigned int verb, unsigned int parm)
+{
+ int err;
+ snd_hda_power_up(codec);
+ mutex_lock(&codec->bus->cmd_mutex);
+ err = codec->bus->ops.command(codec, nid, direct, verb, parm);
+ if (!err) {
+ struct hda_cache_head *c;
+ u32 key = build_cmd_cache_key(nid, verb);
+ c = get_alloc_hash(&codec->cmd_cache, key);
+ if (c)
+ c->val = parm;
+ }
+ mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
+ return err;
+}
+
+/* resume the all commands from the cache */
+void snd_hda_codec_resume_cache(struct hda_codec *codec)
+{
+ struct hda_cache_head *buffer = codec->cmd_cache.buffer;
+ int i;
+
+ for (i = 0; i < codec->cmd_cache.size; i++, buffer++) {
+ u32 key = buffer->key;
+ if (!key)
+ continue;
+ snd_hda_codec_write(codec, get_cmd_cache_nid(key), 0,
+ get_cmd_cache_cmd(key), buffer->val);
+ }
+}
+
+/**
+ * snd_hda_sequence_write_cache - sequence writes with caching
+ * @codec: the HDA codec
+ * @seq: VERB array to send
+ *
+ * Send the commands sequentially from the given array.
+ * Thte commands are recorded on cache for power-save and resume.
+ * The array must be terminated with NID=0.
+ */
+void snd_hda_sequence_write_cache(struct hda_codec *codec,
+ const struct hda_verb *seq)
+{
+ for (; seq->nid; seq++)
+ snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb,
+ seq->param);
+}
+#endif /* SND_HDA_NEEDS_RESUME */
/*
* set power state of the codec
@@ -1345,23 +1622,86 @@ int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec,
static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
- hda_nid_t nid, nid_start;
- int nodes;
+ hda_nid_t nid;
+ int i;
snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE,
power_state);
- nodes = snd_hda_get_sub_nodes(codec, fg, &nid_start);
- for (nid = nid_start; nid < nodes + nid_start; nid++) {
- if (get_wcaps(codec, nid) & AC_WCAP_POWER)
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ if (get_wcaps(codec, nid) & AC_WCAP_POWER) {
+ unsigned int pincap;
+ /*
+ * don't power down the widget if it controls eapd
+ * and EAPD_BTLENABLE is set.
+ */
+ pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+ if (pincap & AC_PINCAP_EAPD) {
+ int eapd = snd_hda_codec_read(codec, nid,
+ 0, AC_VERB_GET_EAPD_BTLENABLE, 0);
+ eapd &= 0x02;
+ if (power_state == AC_PWRST_D3 && eapd)
+ continue;
+ }
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_POWER_STATE,
power_state);
+ }
}
- if (power_state == AC_PWRST_D0)
+ if (power_state == AC_PWRST_D0) {
+ unsigned long end_time;
+ int state;
msleep(10);
+ /* wait until the codec reachs to D0 */
+ end_time = jiffies + msecs_to_jiffies(500);
+ do {
+ state = snd_hda_codec_read(codec, fg, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+ if (state == power_state)
+ break;
+ msleep(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * call suspend and power-down; used both from PM and power-save
+ */
+static void hda_call_codec_suspend(struct hda_codec *codec)
+{
+ if (codec->patch_ops.suspend)
+ codec->patch_ops.suspend(codec, PMSG_SUSPEND);
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D3);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+ codec->power_on = 0;
+ codec->power_transition = 0;
+#endif
+}
+
+/*
+ * kick up codec; used both from PM and power-save
+ */
+static void hda_call_codec_resume(struct hda_codec *codec)
+{
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D0);
+ if (codec->patch_ops.resume)
+ codec->patch_ops.resume(codec);
+ else {
+ if (codec->patch_ops.init)
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ }
}
+#endif /* SND_HDA_NEEDS_RESUME */
/**
@@ -1376,28 +1716,24 @@ int __devinit snd_hda_build_controls(struct hda_bus *bus)
{
struct hda_codec *codec;
- /* build controls */
list_for_each_entry(codec, &bus->codec_list, list) {
- int err;
- if (!codec->patch_ops.build_controls)
- continue;
- err = codec->patch_ops.build_controls(codec);
- if (err < 0)
- return err;
- }
-
- /* initialize */
- list_for_each_entry(codec, &bus->codec_list, list) {
- int err;
+ int err = 0;
+ /* fake as if already powered-on */
+ hda_keep_power_on(codec);
+ /* then fire up */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D0);
- if (!codec->patch_ops.init)
- continue;
- err = codec->patch_ops.init(codec);
+ /* continue to initialize... */
+ if (codec->patch_ops.init)
+ err = codec->patch_ops.init(codec);
+ if (!err && codec->patch_ops.build_controls)
+ err = codec->patch_ops.build_controls(codec);
+ snd_hda_power_down(codec);
if (err < 0)
return err;
}
+
return 0;
}
@@ -1789,9 +2125,9 @@ int __devinit snd_hda_build_pcms(struct hda_bus *bus)
*
* If no entries are matching, the function returns a negative value.
*/
-int __devinit snd_hda_check_board_config(struct hda_codec *codec,
- int num_configs, const char **models,
- const struct snd_pci_quirk *tbl)
+int snd_hda_check_board_config(struct hda_codec *codec,
+ int num_configs, const char **models,
+ const struct snd_pci_quirk *tbl)
{
if (codec->bus->modelname && models) {
int i;
@@ -1841,10 +2177,9 @@ int __devinit snd_hda_check_board_config(struct hda_codec *codec,
*
* Returns 0 if successful, or a negative error code.
*/
-int __devinit snd_hda_add_new_ctls(struct hda_codec *codec,
- struct snd_kcontrol_new *knew)
+int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
{
- int err;
+ int err;
for (; knew->name; knew++) {
struct snd_kcontrol *kctl;
@@ -1867,6 +2202,93 @@ int __devinit snd_hda_add_new_ctls(struct hda_codec *codec,
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+ unsigned int power_state);
+
+static void hda_power_work(struct work_struct *work)
+{
+ struct hda_codec *codec =
+ container_of(work, struct hda_codec, power_work.work);
+
+ if (!codec->power_on || codec->power_count) {
+ codec->power_transition = 0;
+ return;
+ }
+
+ hda_call_codec_suspend(codec);
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+}
+
+static void hda_keep_power_on(struct hda_codec *codec)
+{
+ codec->power_count++;
+ codec->power_on = 1;
+}
+
+void snd_hda_power_up(struct hda_codec *codec)
+{
+ codec->power_count++;
+ if (codec->power_on || codec->power_transition)
+ return;
+
+ codec->power_on = 1;
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+ hda_call_codec_resume(codec);
+ cancel_delayed_work(&codec->power_work);
+ codec->power_transition = 0;
+}
+
+void snd_hda_power_down(struct hda_codec *codec)
+{
+ --codec->power_count;
+ if (!codec->power_on || codec->power_count || codec->power_transition)
+ return;
+ if (power_save) {
+ codec->power_transition = 1; /* avoid reentrance */
+ schedule_delayed_work(&codec->power_work,
+ msecs_to_jiffies(power_save * 1000));
+ }
+}
+
+int snd_hda_check_amp_list_power(struct hda_codec *codec,
+ struct hda_loopback_check *check,
+ hda_nid_t nid)
+{
+ struct hda_amp_list *p;
+ int ch, v;
+
+ if (!check->amplist)
+ return 0;
+ for (p = check->amplist; p->nid; p++) {
+ if (p->nid == nid)
+ break;
+ }
+ if (!p->nid)
+ return 0; /* nothing changed */
+
+ for (p = check->amplist; p->nid; p++) {
+ for (ch = 0; ch < 2; ch++) {
+ v = snd_hda_codec_amp_read(codec, p->nid, ch, p->dir,
+ p->idx);
+ if (!(v & HDA_AMP_MUTE) && v > 0) {
+ if (!check->power_on) {
+ check->power_on = 1;
+ snd_hda_power_up(codec);
+ }
+ return 1;
+ }
+ }
+ }
+ if (check->power_on) {
+ check->power_on = 0;
+ snd_hda_power_down(codec);
+ }
+ return 0;
+}
+#endif
/*
* Channel mode helper
@@ -1913,12 +2335,12 @@ int snd_hda_ch_mode_put(struct hda_codec *codec,
mode = ucontrol->value.enumerated.item[0];
snd_assert(mode < num_chmodes, return -EINVAL);
- if (*max_channelsp == chmode[mode].channels && !codec->in_resume)
+ if (*max_channelsp == chmode[mode].channels)
return 0;
/* change the current channel setting */