Commit e349792a authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (290 commits)
  ALSA: pcm - Update document about xrun_debug proc file
  ALSA: lx6464es - support standard alsa module parameters
  ALSA: snd_usb_caiaq: set mixername
  ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
  ALSA: use card device as parent for jack input-devices
  ALSA: sound/ps3: Correct existing and add missing annotations
  ALSA: sound/ps3: Restructure driver source
  ALSA: sound/ps3: Fix checkpatch issues
  ASoC: Fix lm4857 control
  ALSA: ctxfi - Clear PCM resources at hw_params and hw_free
  ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callbacks
  ALSA: ctxfi - Add missing start check in atc_pcm_playback_start()
  ALSA: ctxfi - Add use_system_timer module option
  ALSA: usb - Add boot quirk for C-Media 6206 USB Audio
  ALSA: ctxfi - Fix wrong model id for UAA
  ALSA: ctxfi - Clean up probe routines
  ALSA: hda - Fix the previous tagra-8ch patch
  ALSA: hda - Add 7.1 support for MSI GX620
  ALSA: pcm - A helper function to compose PCM stream name for debug prints
  ALSA: emu10k1 - Fix minimum periods for efx playback
  ...
parents 6d214918 e3f86d3d
......@@ -460,6 +460,25 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
The power-management is supported.
Module snd-ctxfi
----------------
Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips)
* Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series
* Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series
* Creative Sound Blaster X-Fi Titanium Professional Audio
* Creative Sound Blaster X-Fi Titanium
* Creative Sound Blaster X-Fi Elite Pro
* Creative Sound Blaster X-Fi Platinum
* Creative Sound Blaster X-Fi Fatal1ty
* Creative Sound Blaster X-Fi XtremeGamer
* Creative Sound Blaster X-Fi XtremeMusic
reference_rate - reference sample rate, 44100 or 48000 (default)
multiple - multiple to ref. sample rate, 1 or 2 (default)
This module supports multiple cards.
Module snd-darla20
------------------
......@@ -925,6 +944,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
* Onkyo SE-90PCI
* Onkyo SE-200PCI
* ESI Juli@
* ESI Maya44
* Hercules Fortissimo IV
* EGO-SYS WaveTerminal 192M
......@@ -933,7 +953,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192,
juli, aureon51, aureon71, universe, ap192, k8x800,
phase22, phase28, ms300, av710, se200pci, se90pci,
fortissimo4, sn25p, WT192M
fortissimo4, sn25p, WT192M, maya44
This module supports multiple cards and autoprobe.
......@@ -1093,6 +1113,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
This module supports multiple cards.
The driver requires the firmware loader support on kernel.
Module snd-lx6464es
-------------------
Module for Digigram LX6464ES boards
This module supports multiple cards.
Module snd-maestro3
-------------------
......@@ -1543,13 +1570,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module snd-sc6000
-----------------
Module for Gallant SC-6000 soundcard.
Module for Gallant SC-6000 soundcard and later models: SC-6600
and SC-7000.
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)
joystick - Enable gameport - 0 = disable (default), 1 = enable
This module supports multiple cards.
......@@ -1859,7 +1888,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
-------------------
Module for sound cards based on the Asus AV100/AV200 chips,
i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), and Essence STX.
i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST
(Deluxe) and Essence STX.
This module supports autoprobe and multiple cards.
......
......@@ -36,6 +36,7 @@ ALC260
acer Acer TravelMate
will Will laptops (PB V7900)
replacer Replacer 672V
favorit100 Maxdata Favorit 100XS
basic fixed pin assignment (old default model)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
......@@ -85,10 +86,11 @@ ALC269
eeepc-p703 ASUS Eeepc P703 P900A
eeepc-p901 ASUS Eeepc P901 S101
fujitsu FSC Amilo
lifebook Fujitsu Lifebook S6420
auto auto-config reading BIOS (default)
ALC662/663
==========
ALC662/663/272
==============
3stack-dig 3-stack (2-channel) with SPDIF
3stack-6ch 3-stack (6-channel)
3stack-6ch-dig 3-stack (6-channel) with SPDIF
......@@ -107,6 +109,9 @@ ALC662/663
asus-mode4 ASUS
asus-mode5 ASUS
asus-mode6 ASUS
dell Dell with ALC272
dell-zm1 Dell ZM1 with ALC272
samsung-nc10 Samsung NC10 mini notebook
auto auto-config reading BIOS (default)
ALC882/885
......@@ -118,6 +123,7 @@ ALC882/885
asus-a7j ASUS A7J
asus-a7m ASUS A7M
macpro MacPro support
mb5 Macbook 5,1
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
......@@ -133,10 +139,12 @@ ALC883/888
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
acer-aspire-4930g Acer Aspire 4930G
acer-aspire-8930g Acer Aspire 8930G
medion Medion Laptops
medion-md2 Medion MD2
targa-dig Targa/MSI
targa-2ch-dig Targs/MSI with 2-channel
targa-2ch-dig Targa/MSI with 2-channel
targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
lenovo-101e Lenovo 101E
lenovo-nb0763 Lenovo NB0763
......@@ -150,6 +158,9 @@ ALC883/888
fujitsu-pi2515 Fujitsu AMILO Pi2515
fujitsu-xa3530 Fujitsu AMILO XA3530
3stack-6ch-intel Intel DG33* boards
asus-p5q ASUS P5Q-EM boards
mb31 MacBook 3,1
sony-vaio-tt Sony VAIO TT
auto auto-config reading BIOS (default)
ALC861/660
......@@ -348,6 +359,7 @@ STAC92HD71B*
hp-m4 HP mini 1000
hp-dv5 HP dv series
hp-hdx HP HDX series
hp-dv4-1222nr HP dv4-1222nr (with LED support)
auto BIOS setup (default)
STAC92HD73*
......
......@@ -88,26 +88,34 @@ card*/pcm*/info
substreams, etc.
card*/pcm*/xrun_debug
This file appears when CONFIG_SND_DEBUG=y.
This shows the status of xrun (= buffer overrun/xrun) debug of
ALSA PCM middle layer, as an integer from 0 to 2. The value
can be changed by writing to this file, such as
# cat 2 > /proc/asound/card0/pcm0p/xrun_debug
When this value is greater than 0, the driver will show the
messages to kernel log when an xrun is detected. The debug
message is shown also when the invalid H/W pointer is detected
at the update of periods (usually called from the interrupt
This file appears when CONFIG_SND_DEBUG=y and
CONFIG_PCM_XRUN_DEBUG=y.
This shows the status of xrun (= buffer overrun/xrun) and
invalid PCM position debug/check of ALSA PCM middle layer.
It takes an integer value, can be changed by writing to this
file, such as
# cat 5 > /proc/asound/card0/pcm0p/xrun_debug
The value consists of the following bit flags:
bit 0 = Enable XRUN/jiffies debug messages
bit 1 = Show stack trace at XRUN / jiffies check
bit 2 = Enable additional jiffies check
When the bit 0 is set, the driver will show the messages to
kernel log when an xrun is detected. The debug message is
shown also when the invalid H/W pointer is detected at the
update of periods (usually called from the interrupt
handler).
When this value is greater than 1, the driver will show the
stack trace additionally. This may help the debugging.
When the bit 1 is set, the driver will show the stack trace
additionally. This may help the debugging.
Since 2.6.30, this option also enables the hwptr check using
Since 2.6.30, this option can enable the hwptr check using
jiffies. This detects spontaneous invalid pointer callback
values, but can be lead to too much corrections for a (mostly
buggy) hardware that doesn't give smooth pointer updates.
This feature is enabled via the bit 2.
card*/pcm*/sub*/info
The general information of this PCM sub-stream.
......
NOTE: The following is the original document of Rainer's patch that the
current maya44 code based on. Some contents might be obsoleted, but I
keep here as reference -- tiwai
----------------------------------------------------------------
STATE OF DEVELOPMENT:
This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
Development is carried out by Rainer Zimmermann (mail@lightshed.de).
ESI provided a sample Maya44 card for the development work.
However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
- playback and capture at all sampling rates
- input/output level
- crossmixing
- line/mic switch
- phantom power switch
- analogue monitor a.k.a bypass
The following functions *should* work, but are not fully tested:
- Channel 3+4 analogue - S/PDIF input switching
- S/PDIF output
- all inputs/outputs on the M/IO/DIO extension card
- internal/external clock selection
*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
Things that do not seem to work:
- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
DRIVER DETAILS:
the following files were added:
pci/ice1724/maya44.c - Maya44 specific code
pci/ice1724/maya44.h
pci/ice1724/ice1724.patch
pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs
include/wm8776.h
Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
wtm.h
vt1720_mobo.h
revo.h
prodigy192.h
pontis.h
phase.h
maya44.h
juli.h
aureon.h
amp.h
envy24ht.h
se.h
prodigy_hifi.h
*I hope this is the correct way to do things.*
SAMPLING RATES:
The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
SOUND DEVICES:
PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
hw:0,0 input - stereo, analog input 1+2
hw:0,0 output - stereo, analog output 1+2
hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
NAMING OF MIXER CONTROLS:
(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
PCM: (digital) output level for channel 1+2
PCM 1: same for channel 3+4
Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
Make sure this is not turned on while any other source is connected to input 1/2.
It might damage the source and/or the maya44 card.
Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
Bypass 1: same for channel 3+4.
Crossmix: cross-mixer from channels 1+2 to channels 3+4
Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
IEC958 Output: switch for S/PDIF output.
This is not supported by the ESI windows driver.
S/PDIF should output the same signal as channel 3+4. [untested!]
Digitial output selectors:
These switches allow a direct digital routing from the ADCs to the DACs.
Each switch determines where the digital input data to one of the DACs comes from.
They are not supported by the ESI windows driver.
For normal operation, they should all be set to "PCM out".
H/W: Output source channel 1
H/W 1: Output source channel 2
H/W 2: Output source channel 3
H/W 3: Output source channel 4
H/W 4 ... H/W 9: unknown function, left in to enable testing.
Possibly some of these control S/PDIF output(s).
If these turn out to be unused, they will go away in later driver versions.
Selectable values for each of the digital output selectors are:
"PCM out" -> DAC output of the corresponding channel (default setting)
"Input 1"...
"Input 4" -> direct routing from ADC output of the selected input channel
--------
Feb 14, 2008
Rainer Zimmermann
mail@lightshed.de
......@@ -62,6 +62,7 @@ Audio DAPM widgets fall into a number of types:-
o Mic - Mic (and optional Jack)
o Line - Line Input/Output (and optional Jack)
o Speaker - Speaker
o Supply - Power or clock supply widget used by other widgets.
o Pre - Special PRE widget (exec before all others)
o Post - Special POST widget (exec after all others)
......
......@@ -4603,7 +4603,8 @@ F: drivers/pcmcia/pxa2xx*
F: drivers/spi/pxa2xx*
F: drivers/usb/gadget/pxa2*
F: include/sound/pxa2xx-lib.h
F: sound/soc/pxa/pxa2xx*
F: sound/arm/pxa*
F: sound/soc/pxa
PXA168 SUPPORT
P: Eric Miao
......@@ -5331,11 +5332,12 @@ P: Liam Girdwood
M: lrg@slimlogic.co.uk
P: Mark Brown
M: broonie@opensource.wolfsonmicro.com
T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
L: alsa-devel@alsa-project.org (subscribers-only)
W: http://alsa-project.org/main/index.php/ASoC
S: Supported
F: sound/soc/
F: include/sound/soc*
SPARC + UltraSPARC (sparc/sparc64)
P: David S. Miller
......
......@@ -28,6 +28,10 @@
#define MPC52xx_PSC_MAXNUM 6
/* Programmable Serial Controller (PSC) status register bits */
#define MPC52xx_PSC_SR_UNEX_RX 0x0001
#define MPC52xx_PSC_SR_DATA_VAL 0x0002
#define MPC52xx_PSC_SR_DATA_OVR 0x0004
#define MPC52xx_PSC_SR_CMDSEND 0x0008
#define MPC52xx_PSC_SR_CDE 0x0080
#define MPC52xx_PSC_SR_RXRDY 0x0100
#define MPC52xx_PSC_SR_RXFULL 0x0200
......@@ -61,6 +65,12 @@
#define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001
/* PSC interrupt status/mask bits */
#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001
#define MPC52xx_PSC_IMR_DATA_VALID 0x0002
#define MPC52xx_PSC_IMR_DATA_OVR 0x0004
#define MPC52xx_PSC_IMR_CMD_SEND 0x0008
#define MPC52xx_PSC_IMR_ERROR 0x0040
#define MPC52xx_PSC_IMR_DEOF 0x0080
#define MPC52xx_PSC_IMR_TXRDY 0x0100
#define MPC52xx_PSC_IMR_RXRDY 0x0200
#define MPC52xx_PSC_IMR_DB 0x0400
......@@ -117,6 +127,7 @@
#define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24)
#define MPC52xx_PSC_SICR_AWR (1 << 30)
#define MPC52xx_PSC_SICR_GENCLK (1 << 23)
#define MPC52xx_PSC_SICR_I2S (1 << 22)
#define MPC52xx_PSC_SICR_CLKPOL (1 << 21)
......
......@@ -1005,6 +1005,7 @@
#define PCI_DEVICE_ID_PLX_PCI200SYN 0x3196
#define PCI_DEVICE_ID_PLX_9030 0x9030
#define PCI_DEVICE_ID_PLX_9050 0x9050
#define PCI_DEVICE_ID_PLX_9056 0x9056
#define PCI_DEVICE_ID_PLX_9080 0x9080
#define PCI_DEVICE_ID_PLX_GTEK_SERIAL2 0xa001
......@@ -1314,6 +1315,13 @@
#define PCI_VENDOR_ID_CREATIVE 0x1102 /* duplicate: ECTIVA */
#define PCI_DEVICE_ID_CREATIVE_EMU10K1 0x0002
#define PCI_DEVICE_ID_CREATIVE_20K1 0x0005
#define PCI_DEVICE_ID_CREATIVE_20K2 0x000b
#define PCI_SUBDEVICE_ID_CREATIVE_SB0760 0x0024
#define PCI_SUBDEVICE_ID_CREATIVE_SB08801 0x0041
#define PCI_SUBDEVICE_ID_CREATIVE_SB08802 0x0042
#define PCI_SUBDEVICE_ID_CREATIVE_SB08803 0x0043
#define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX 0x6000
#define PCI_VENDOR_ID_ECTIVA 0x1102 /* duplicate: CREATIVE */
#define PCI_DEVICE_ID_ECTIVA_EV1938 0x8938
......@@ -1847,6 +1855,10 @@
#define PCI_SUBDEVICE_ID_HYPERCOPE_METRO 0x0107
#define PCI_SUBDEVICE_ID_HYPERCOPE_CHAMP2 0x0108
#define PCI_VENDOR_ID_DIGIGRAM 0x1369
#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM 0xc001
#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM 0xc002
#define PCI_VENDOR_ID_KAWASAKI 0x136b
#define PCI_DEVICE_ID_MCHIP_KL5A72002 0xff01
......
......@@ -255,6 +255,7 @@ typedef int __bitwise snd_pcm_subformat_t;
#define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000 /* only half duplex */
#define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */
#define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */
#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */
typedef int __bitwise snd_pcm_state_t;
#define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */
......
......@@ -300,19 +300,10 @@ int snd_card_create(int idx, const char *id,
struct module *module, int extra_size,
struct snd_card **card_ret);
static inline __deprecated
struct snd_card *snd_card_new(int idx, const char *id,
struct module *module, int extra_size)
{
struct snd_card *card;
if (snd_card_create(idx, id, module, extra_size, &card) < 0)
return NULL;
return card;
}
int snd_card_disconnect(struct snd_card *card);
int snd_card_free(struct snd_card *card);
int snd_card_free_when_closed(struct snd_card *card);
void snd_card_set_id(struct snd_card *card, const char *id);
int snd_card_register(struct snd_card *card);
int snd_card_info_init(void);
int snd_card_info_done(void);
......
#warning "This file is deprecated"
......@@ -98,6 +98,7 @@ struct snd_pcm_ops {
#define SNDRV_PCM_IOCTL1_INFO 1
#define SNDRV_PCM_IOCTL1_CHANNEL_INFO 2
#define SNDRV_PCM_IOCTL1_GSTATE 3
#define SNDRV_PCM_IOCTL1_FIFO_SIZE 4
#define SNDRV_PCM_TRIGGER_STOP 0
#define SNDRV_PCM_TRIGGER_START 1
......@@ -270,6 +271,7 @@ struct snd_pcm_runtime {
snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */
snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */
unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */
snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */
/* -- HW params -- */
snd_pcm_access_t access; /* access mode */
......@@ -486,80 +488,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream);
void snd_pcm_vma_notify_data(void *client, void *data);
int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area);
#if BITS_PER_LONG >= 64
static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem)
{
*rem = *n % div;
*n /= div;
}
#elif defined(i386)
static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem)
{
u_int32_t low, high;
low = *n & 0xffffffff;
high = *n >> 32;
if (high) {
u_int32_t high1 = high % div;
high /= div;
asm("divl %2":"=a" (low), "=d" (*rem):"rm" (div), "a" (low), "d" (high1));
*n = (u_int64_t)high << 32 | low;
} else {
*n = low / div;
*rem = low % div;
}
}
#else
static inline void divl(u_int32_t high, u_int32_t low,
u_int32_t div,
u_int32_t *q, u_int32_t *r)
{
u_int64_t n = (u_int64_t)high << 32 | low;
u_int64_t d = (u_int64_t)div << 31;
u_int32_t q1 = 0;
int c = 32;
while (n > 0xffffffffU) {
q1 <<= 1;
if (n >= d) {
n -= d;
q1 |= 1;
}
d >>= 1;
c--;
}
q1 <<= c;
if (n) {
low = n;
*q = q1 | (low / div);
*r = low % div;
} else {
*r = 0;
*q = q1;
}
return;
}
static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem)
{
u_int32_t low, high;
low = *n & 0xffffffff;
high = *n >> 32;
if (high) {
u_int32_t high1 = high % div;
u_int32_t low1 = low;
high /= div;
divl(high1, low1, div, &low, rem);
*n = (u_int64_t)high << 32 | low;
} else {
*n = low / div;
*rem = low % div;
}
}
#endif
/*
* PCM library
*/
......
......@@ -44,24 +44,6 @@ struct snd_pcm_substream;
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
/*
* DAI Left/Right Clocks.
*
* Specifies whether the DAI can support different samples for similtanious
* playback and capture. This usually requires a seperate physical frame
* clock for playback and capture.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*
* Time Division Multiplexing. Allows PCM data to be multplexed with other
* data on the DAI.
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* DAI hardware signal inversions.
*
......@@ -96,6 +78,10 @@ struct snd_pcm_substream;
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
SNDRV_PCM_FMTBIT_S32_LE |\
SNDRV_PCM_FMTBIT_S32_BE)
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
......@@ -208,6 +194,7 @@ struct snd_soc_dai {
/* DAI capabilities */
struct snd_soc_pcm_stream capture;