Re: [RFC: 2.6 patch] powerpc: remove the unused HTDMSOUND driver

From: Marcelo Tosatti
Date: Tue Mar 27 2007 - 12:03:25 EST

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In reply to: Adrian Bunk: "[RFC: 2.6 patch] powerpc: remove the unused HTDMSOUND driver"
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Dan,

Shall this driver be removed?

On Sun, Mar 25, 2007 at 04:58:38PM +0200, Adrian Bunk wrote:
> Recently, someone fixed a syntax error in the HTDMSOUND driver
> introduced 4 years ago.
>
> Unfortunately not by trying to compile this driver for his hardware but
> by code inspection - which seems to be a strong indication that there
> are no users left for this OSS sound driver.
>
> This patch therefore removes it.
>
> Signed-off-by: Adrian Bunk <bunk@xxxxxxxxx>
>
> ---
>
> This patch was already sent on:
> - 7 Mar 2007
>
> arch/ppc/8xx_io/Kconfig | 4
> arch/ppc/8xx_io/Makefile | 1
> arch/ppc/8xx_io/cs4218.h | 166 -
> arch/ppc/8xx_io/cs4218_tdm.c | 2833 --------------------------------
> arch/ppc/platforms/rpxclassic.h | 4
> arch/ppc/platforms/rpxhiox.h | 41
> arch/ppc/platforms/rpxlite.h | 4
> arch/ppc/syslib/m8xx_setup.c | 2
> 8 files changed, 1 insertion(+), 3054 deletions(-)
>
> --- linux-2.6.21-rc2-mm1/arch/ppc/8xx_io/Kconfig.old 2007-03-06 06:47:04.000000000 +0100
> +++ linux-2.6.21-rc2-mm1/arch/ppc/8xx_io/Kconfig 2007-03-06 06:47:16.000000000 +0100
> @@ -74,10 +74,6 @@
> Allocate large buffers for MPC8xx Ethernet. Increases throughput
> and decreases the likelihood of dropped packets, but costs memory.
>
> -config HTDMSOUND
> - bool "Embedded Planet HIOX Audio"
> - depends on SOUND=y
> -
> # This doesn't really belong here, but it is convenient to ask
> # 8xx specific questions.
> comment "Generic MPC8xx Options"
> --- linux-2.6.21-rc2-mm1/arch/ppc/8xx_io/Makefile.old 2007-03-06 06:47:23.000000000 +0100
> +++ linux-2.6.21-rc2-mm1/arch/ppc/8xx_io/Makefile 2007-03-06 06:47:30.000000000 +0100
> @@ -7,4 +7,3 @@
> obj-$(CONFIG_FEC_ENET) += fec.o
> obj-$(CONFIG_SCC_ENET) += enet.o
> obj-$(CONFIG_UCODE_PATCH) += micropatch.o
> -obj-$(CONFIG_HTDMSOUND) += cs4218_tdm.o
> --- linux-2.6.21-rc2-mm1/arch/ppc/platforms/rpxlite.h.old 2007-03-06 06:49:05.000000000 +0100
> +++ linux-2.6.21-rc2-mm1/arch/ppc/platforms/rpxlite.h 2007-03-06 06:49:41.000000000 +0100
> @@ -57,10 +57,6 @@
> #define BCSR1_PCVCTL6 ((uint)0x00020000)
> #define BCSR1_PCVCTL7 ((uint)0x00010000)
>
> -#if defined(CONFIG_HTDMSOUND)
> -#include <platforms/rpxhiox.h>
> -#endif
> -
> /* define IO_BASE for pcmcia */
> #define _IO_BASE 0x80000000
> #define _IO_BASE_SIZE 0x1000
> --- linux-2.6.21-rc2-mm1/arch/ppc/platforms/rpxclassic.h.old 2007-03-06 06:49:51.000000000 +0100
> +++ linux-2.6.21-rc2-mm1/arch/ppc/platforms/rpxclassic.h 2007-03-06 06:49:56.000000000 +0100
> @@ -69,10 +69,6 @@
> #define BCSR2_QSPACESEL ((uint)0x00004000)
> #define BCSR2_FETHLEDMODE ((uint)0x00000800) /* CLLF */
>
> -#if defined(CONFIG_HTDMSOUND)
> -#include <platforms/rpxhiox.h>
> -#endif
> -
> /* define IO_BASE for pcmcia, CLLF only */
> #if !defined(CONFIG_PCI)
> #define _IO_BASE 0x80000000
> --- linux-2.6.21-rc2-mm1/arch/ppc/syslib/m8xx_setup.c.old 2007-03-06 06:50:43.000000000 +0100
> +++ linux-2.6.21-rc2-mm1/arch/ppc/syslib/m8xx_setup.c 2007-03-06 06:50:59.000000000 +0100
> @@ -413,7 +413,7 @@
> io_block_mapping(_IO_BASE,_IO_BASE,_IO_BASE_SIZE, _PAGE_IO);
> #endif
> #endif
> -#if defined(CONFIG_HTDMSOUND) || defined(CONFIG_RPXTOUCH) || defined(CONFIG_FB_RPX)
> +#if defined(CONFIG_RPXTOUCH) || defined(CONFIG_FB_RPX)
> io_block_mapping(HIOX_CSR_ADDR, HIOX_CSR_ADDR, HIOX_CSR_SIZE, _PAGE_IO);
> #endif
> #ifdef CONFIG_FADS
> --- linux-2.6.21-rc2-mm1/arch/ppc/platforms/rpxhiox.h 2007-02-04 19:44:54.000000000 +0100
> +++ /dev/null 2006-09-19 00:45:31.000000000 +0200
> @@ -1,41 +0,0 @@
> -/*
> - * The Embedded Planet HIOX expansion card definitions.
> - * There were a few different versions of these cards, but only
> - * the one that escaped real production is defined here.
> - *
> - * Copyright (c) 2000 Dan Malek (dmalek@xxxxxxx)
> - */
> -#ifndef __MACH_RPX_HIOX_DEFS
> -#define __MACH_RPX_HIOX_DEFS
> -
> -#define HIOX_CSR_ADDR ((uint)0xfac00000)
> -#define HIOX_CSR_SIZE ((uint)(4 * 1024))
> -#define HIOX_CSR0_ADDR HIOX_CSR_ADDR
> -#define HIOX_CSR4_ADDR ((uint)0xfac00004)
> -
> -#define HIOX_CSR0_DEFAULT ((uint)0x380f3c00)
> -#define HIOX_CSR0_ENSCC2 ((uint)0x80000000)
> -#define HIOX_CSR0_ENSMC2 ((uint)0x04000000)
> -#define HIOX_CSR0_ENVDOCLK ((uint)0x02000000)
> -#define HIOX_CSR0_VDORST_HL ((uint)0x01000000)
> -#define HIOX_CSR0_RS232SEL ((uint)0x0000c000)
> -#define HIOX_CSR0_SCC3SEL ((uint)0x0000c000)
> -#define HIOX_CSR0_SMC1SEL ((uint)0x00008000)
> -#define HIOX_CSR0_SCC1SEL ((uint)0x00004000)
> -#define HIOX_CSR0_ENTOUCH ((uint)0x00000080)
> -#define HIOX_CSR0_PDOWN100 ((uint)0x00000060)
> -#define HIOX_CSR0_PDOWN10 ((uint)0x00000040)
> -#define HIOX_CSR0_PDOWN1 ((uint)0x00000020)
> -#define HIOX_CSR0_TSELSPI ((uint)0x00000010)
> -#define HIOX_CSR0_TIRQSTAT ((uint)0x00000008)
> -#define HIOX_CSR4_DEFAULT ((uint)0x00000000)
> -#define HIOX_CSR4_ENTIRQ2 ((uint)0x20000000)
> -#define HIOX_CSR4_ENTIRQ3 ((uint)0x10000000)
> -#define HIOX_CSR4_ENAUDIO ((uint)0x00000080)
> -#define HIOX_CSR4_RSTAUDIO ((uint)0x00000040) /* 0 == reset */
> -#define HIOX_CSR4_AUDCLKHI ((uint)0x00000020)
> -#define HIOX_CSR4_AUDSPISEL ((uint)0x00000010)
> -#define HIOX_CSR4_AUDIRQSTAT ((uint)0x00000008)
> -#define HIOX_CSR4_AUDCLKSEL ((uint)0x00000007)
> -
> -#endif
> --- linux-2.6.21-rc2-mm1/arch/ppc/8xx_io/cs4218.h 2007-02-04 19:44:54.000000000 +0100
> +++ /dev/null 2006-09-19 00:45:31.000000000 +0200
> @@ -1,166 +0,0 @@
> -#ifndef _cs4218_h_
> -/*
> - * Hacked version of linux/drivers/sound/dmasound/dmasound.h
> - *
> - *
> - * Minor numbers for the sound driver.
> - *
> - * Unfortunately Creative called the codec chip of SB as a DSP. For this
> - * reason the /dev/dsp is reserved for digitized audio use. There is a
> - * device for true DSP processors but it will be called something else.
> - * In v3.0 it's /dev/sndproc but this could be a temporary solution.
> - */
> -#define _cs4218_h_
> -
> -#include <linux/types.h>
> -
> -#define SND_NDEVS 256 /* Number of supported devices */
> -#define SND_DEV_CTL 0 /* Control port /dev/mixer */
> -#define SND_DEV_SEQ 1 /* Sequencer output /dev/sequencer (FM
> - synthesizer and MIDI output) */
> -#define SND_DEV_MIDIN 2 /* Raw midi access */
> -#define SND_DEV_DSP 3 /* Digitized voice /dev/dsp */
> -#define SND_DEV_AUDIO 4 /* Sparc compatible /dev/audio */
> -#define SND_DEV_DSP16 5 /* Like /dev/dsp but 16 bits/sample */
> -#define SND_DEV_STATUS 6 /* /dev/sndstat */
> -/* #7 not in use now. Was in 2.4. Free for use after v3.0. */
> -#define SND_DEV_SEQ2 8 /* /dev/sequencer, level 2 interface */
> -#define SND_DEV_SNDPROC 9 /* /dev/sndproc for programmable devices */
> -#define SND_DEV_PSS SND_DEV_SNDPROC
> -
> -/* switch on various prinks */
> -#define DEBUG_DMASOUND 1
> -
> -#define MAX_AUDIO_DEV 5
> -#define MAX_MIXER_DEV 4
> -#define MAX_SYNTH_DEV 3
> -#define MAX_MIDI_DEV 6
> -#define MAX_TIMER_DEV 3
> -
> -#define MAX_CATCH_RADIUS 10
> -
> -#define le2be16(x) (((x)<<8 & 0xff00) | ((x)>>8 & 0x00ff))
> -#define le2be16dbl(x) (((x)<<8 & 0xff00ff00) | ((x)>>8 & 0x00ff00ff))
> -
> -#define IOCTL_IN(arg, ret) \
> - do { int error = get_user(ret, (int *)(arg)); \
> - if (error) return error; \
> - } while (0)
> -#define IOCTL_OUT(arg, ret) ioctl_return((int *)(arg), ret)
> -
> -static inline int ioctl_return(int *addr, int value)
> -{
> - return value < 0 ? value : put_user(value, addr);
> -}
> -
> -#define HAS_RECORD
> -
> - /*
> - * Initialization
> - */
> -
> -/* description of the set-up applies to either hard or soft settings */
> -
> -typedef struct {
> - int format; /* AFMT_* */
> - int stereo; /* 0 = mono, 1 = stereo */
> - int size; /* 8/16 bit*/
> - int speed; /* speed */
> -} SETTINGS;
> -
> - /*
> - * Machine definitions
> - */
> -
> -typedef struct {
> - const char *name;
> - const char *name2;
> - void (*open)(void);
> - void (*release)(void);
> - void *(*dma_alloc)(unsigned int, gfp_t);
> - void (*dma_free)(void *, unsigned int);
> - int (*irqinit)(void);
> -#ifdef MODULE
> - void (*irqcleanup)(void);
> -#endif
> - void (*init)(void);
> - void (*silence)(void);
> - int (*setFormat)(int);
> - int (*setVolume)(int);
> - int (*setBass)(int);
> - int (*setTreble)(int);
> - int (*setGain)(int);
> - void (*play)(void);
> - void (*record)(void); /* optional */
> - void (*mixer_init)(void); /* optional */
> - int (*mixer_ioctl)(u_int, u_long); /* optional */
> - int (*write_sq_setup)(void); /* optional */
> - int (*read_sq_setup)(void); /* optional */
> - int (*sq_open)(mode_t); /* optional */
> - int (*state_info)(char *, size_t); /* optional */
> - void (*abort_read)(void); /* optional */
> - int min_dsp_speed;
> - int max_dsp_speed;
> - int version ;
> - int hardware_afmts ; /* OSS says we only return h'ware info */
> - /* when queried via SNDCTL_DSP_GETFMTS */
> - int capabilities ; /* low-level reply to SNDCTL_DSP_GETCAPS */
> - SETTINGS default_hard ; /* open() or init() should set something valid */
> - SETTINGS default_soft ; /* you can make it look like old OSS, if you want to */
> -} MACHINE;
> -
> - /*
> - * Low level stuff
> - */
> -
> -typedef struct {
> - ssize_t (*ct_ulaw)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_alaw)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_s8)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_u8)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_s16be)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_u16be)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_s16le)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> - ssize_t (*ct_u16le)(const u_char *, size_t, u_char *, ssize_t *, ssize_t);
> -} TRANS;
> -
> -
> - /*
> - * Sound queue stuff, the heart of the driver
> - */
> -
> -struct sound_queue {
> - /* buffers allocated for this queue */
> - int numBufs; /* real limits on what the user can have */
> - int bufSize; /* in bytes */
> - char **buffers;
> -
> - /* current parameters */
> - int locked ; /* params cannot be modified when != 0 */
> - int user_frags ; /* user requests this many */
> - int user_frag_size ; /* of this size */
> - int max_count; /* actual # fragments <= numBufs */
> - int block_size; /* internal block size in bytes */
> - int max_active; /* in-use fragments <= max_count */
> -
> - /* it shouldn't be necessary to declare any of these volatile */
> - int front, rear, count;
> - int rear_size;
> - /*
> - * The use of the playing field depends on the hardware
> - *
> - * Atari, PMac: The number of frames that are loaded/playing
> - *
> - * Amiga: Bit 0 is set: a frame is loaded
> - * Bit 1 is set: a frame is playing
> - */
> - int active;
> - wait_queue_head_t action_queue, open_queue, sync_queue;
> - int open_mode;
> - int busy, syncing, xruns, died;
> -};
> -
> -#define SLEEP(queue) interruptible_sleep_on_timeout(&queue, HZ)
> -#define WAKE_UP(queue) (wake_up_interruptible(&queue))
> -
> -#endif /* _cs4218_h_ */
> --- linux-2.6.21-rc2-mm1/arch/ppc/8xx_io/cs4218_tdm.c 2007-03-02 20:14:59.000000000 +0100
> +++ /dev/null 2006-09-19 00:45:31.000000000 +0200
> @@ -1,2833 +0,0 @@
> -
> -/* This is a modified version of linux/drivers/sound/dmasound.c to
> - * support the CS4218 codec on the 8xx TDM port. Thanks to everyone
> - * that contributed to the dmasound software (which includes me :-).
> - *
> - * The CS4218 is configured in Mode 4, sub-mode 0. This provides
> - * left/right data only on the TDM port, as a 32-bit word, per frame
> - * pulse. The control of the CS4218 is provided by some other means,
> - * like the SPI port.
> - * Dan Malek (dmalek@xxxxxxx)
> - */
> -
> -#include <linux/module.h>
> -#include <linux/sched.h>
> -#include <linux/timer.h>
> -#include <linux/major.h>
> -#include <linux/fcntl.h>
> -#include <linux/errno.h>
> -#include <linux/mm.h>
> -#include <linux/slab.h>
> -#include <linux/sound.h>
> -#include <linux/init.h>
> -#include <linux/delay.h>
> -
> -#include <asm/system.h>
> -#include <asm/irq.h>
> -#include <asm/pgtable.h>
> -#include <asm/uaccess.h>
> -#include <asm/io.h>
> -
> -/* Should probably do something different with this path name.....
> - * Actually, I should just stop using it...
> - */
> -#include "cs4218.h"
> -#include <linux/soundcard.h>
> -
> -#include <asm/mpc8xx.h>
> -#include <asm/8xx_immap.h>
> -#include <asm/commproc.h>
> -
> -#define DMASND_CS4218 5
> -
> -#define MAX_CATCH_RADIUS 10
> -#define MIN_BUFFERS 4
> -#define MIN_BUFSIZE 4
> -#define MAX_BUFSIZE 128
> -
> -#define HAS_8BIT_TABLES
> -
> -static int sq_unit = -1;
> -static int mixer_unit = -1;
> -static int state_unit = -1;
> -static int irq_installed = 0;
> -static char **sound_buffers = NULL;
> -static char **sound_read_buffers = NULL;
> -
> -static DEFINE_SPINLOCK(cs4218_lock);
> -
> -/* Local copies of things we put in the control register. Output
> - * volume, like most codecs is really attenuation.
> - */
> -static int cs4218_rate_index;
> -
> -/*
> - * Stuff for outputting a beep. The values range from -327 to +327
> - * so we can multiply by an amplitude in the range 0..100 to get a
> - * signed short value to put in the output buffer.
> - */
> -static short beep_wform[256] = {
> - 0, 40, 79, 117, 153, 187, 218, 245,
> - 269, 288, 304, 316, 323, 327, 327, 324,
> - 318, 310, 299, 288, 275, 262, 249, 236,
> - 224, 213, 204, 196, 190, 186, 183, 182,
> - 182, 183, 186, 189, 192, 196, 200, 203,
> - 206, 208, 209, 209, 209, 207, 204, 201,
> - 197, 193, 188, 183, 179, 174, 170, 166,
> - 163, 161, 160, 159, 159, 160, 161, 162,
> - 164, 166, 168, 169, 171, 171, 171, 170,
> - 169, 167, 163, 159, 155, 150, 144, 139,
> - 133, 128, 122, 117, 113, 110, 107, 105,
> - 103, 103, 103, 103, 104, 104, 105, 105,
> - 105, 103, 101, 97, 92, 86, 78, 68,
> - 58, 45, 32, 18, 3, -11, -26, -41,
> - -55, -68, -79, -88, -95, -100, -102, -102,
> - -99, -93, -85, -75, -62, -48, -33, -16,
> - 0, 16, 33, 48, 62, 75, 85, 93,
> - 99, 102, 102, 100, 95, 88, 79, 68,
> - 55, 41, 26, 11, -3, -18, -32, -45,
> - -58, -68, -78, -86, -92, -97, -101, -103,
> - -105, -105, -105, -104, -104, -103, -103, -103,
> - -103, -105, -107, -110, -113, -117, -122, -128,
> - -133, -139, -144, -150, -155, -159, -163, -167,
> - -169, -170, -171, -171, -171, -169, -168, -166,
> - -164, -162, -161, -160, -159, -159, -160, -161,
> - -163, -166, -170, -174, -179, -183, -188, -193,
> - -197, -201, -204, -207, -209, -209, -209, -208,
> - -206, -203, -200, -196, -192, -189, -186, -183,
> - -182, -182, -183, -186, -190, -196, -204, -213,
> - -224, -236, -249, -262, -275, -288, -299, -310,
> - -318, -324, -327, -327, -323, -316, -304, -288,
> - -269, -245, -218, -187, -153, -117, -79, -40,
> -};
> -
> -#define BEEP_SPEED 5 /* 22050 Hz sample rate */
> -#define BEEP_BUFLEN 512
> -#define BEEP_VOLUME 15 /* 0 - 100 */
> -
> -static int beep_volume = BEEP_VOLUME;
> -static int beep_playing = 0;
> -static int beep_state = 0;
> -static short *beep_buf;
> -static void (*orig_mksound)(unsigned int, unsigned int);
> -
> -/* This is found someplace else......I guess in the keyboard driver
> - * we don't include.
> - */
> -static void (*kd_mksound)(unsigned int, unsigned int);
> -
> -static int catchRadius = 0;
> -static int numBufs = 4, bufSize = 32;
> -static int numReadBufs = 4, readbufSize = 32;
> -
> -
> -/* TDM/Serial transmit and receive buffer descriptors.
> -*/
> -static volatile cbd_t *rx_base, *rx_cur, *tx_base, *tx_cur;
> -
> -module_param(catchRadius, int, 0);
> -module_param(numBufs, int, 0);
> -module_param(bufSize, int, 0);
> -module_param(numreadBufs, int, 0);
> -module_param(readbufSize, int, 0);
> -
> -#define arraysize(x) (sizeof(x)/sizeof(*(x)))
> -#define le2be16(x) (((x)<<8 & 0xff00) | ((x)>>8 & 0x00ff))
> -#define le2be16dbl(x) (((x)<<8 & 0xff00ff00) | ((x)>>8 & 0x00ff00ff))
> -
> -#define IOCTL_IN(arg, ret) \
> - do { int error = get_user(ret, (int *)(arg)); \
> - if (error) return error; \
> - } while (0)
> -#define IOCTL_OUT(arg, ret) ioctl_return((int *)(arg), ret)
> -
> -/* CS4218 serial port control in mode 4.
> -*/
> -#define CS_INTMASK ((uint)0x40000000)
> -#define CS_DO1 ((uint)0x20000000)
> -#define CS_LATTEN ((uint)0x1f000000)
> -#define CS_RATTEN ((uint)0x00f80000)
> -#define CS_MUTE ((uint)0x00040000)
> -#define CS_ISL ((uint)0x00020000)
> -#define CS_ISR ((uint)0x00010000)
> -#define CS_LGAIN ((uint)0x0000f000)
> -#define CS_RGAIN ((uint)0x00000f00)
> -
> -#define CS_LATTEN_SET(X) (((X) & 0x1f) << 24)
> -#define CS_RATTEN_SET(X) (((X) & 0x1f) << 19)
> -#define CS_LGAIN_SET(X) (((X) & 0x0f) << 12)
> -#define CS_RGAIN_SET(X) (((X) & 0x0f) << 8)
> -
> -#define CS_LATTEN_GET(X) (((X) >> 24) & 0x1f)
> -#define CS_RATTEN_GET(X) (((X) >> 19) & 0x1f)
> -#define CS_LGAIN_GET(X) (((X) >> 12) & 0x0f)
> -#define CS_RGAIN_GET(X) (((X) >> 8) & 0x0f)
> -
> -/* The control register is effectively write only. We have to keep a copy
> - * of what we write.
> - */
> -static uint cs4218_control;
> -
> -/* A place to store expanding information.
> -*/
> -static int expand_bal;
> -static int expand_data;
> -
> -/* Since I can't make the microcode patch work for the SPI, I just
> - * clock the bits using software.
> - */
> -static void sw_spi_init(void);
> -static void sw_spi_io(u_char *obuf, u_char *ibuf, uint bcnt);
> -static uint cs4218_ctl_write(uint ctlreg);
> -
> -/*** Some low level helpers **************************************************/
> -
> -/* 16 bit mu-law */
> -
> -static short ulaw2dma16[] = {
> - -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956,
> - -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764,
> - -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412,
> - -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316,
> - -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140,
> - -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092,
> - -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004,
> - -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980,
> - -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436,
> - -1372, -1308, -1244, -1180, -1116, -1052, -988, -924,
> - -876, -844, -812, -780, -748, -716, -684, -652,
> - -620, -588, -556, -524, -492, -460, -428, -396,
> - -372, -356, -340, -324, -308, -292, -276, -260,
> - -244, -228, -212, -196, -180, -164, -148, -132,
> - -120, -112, -104, -96, -88, -80, -72, -64,
> - -56, -48, -40, -32, -24, -16, -8, 0,
> - 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956,
> - 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764,
> - 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412,
> - 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316,
> - 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140,
> - 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092,
> - 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004,
> - 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980,
> - 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436,
> - 1372, 1308, 1244, 1180, 1116, 1052, 988, 924,
> - 876, 844, 812, 780, 748, 716, 684, 652,
> - 620, 588, 556, 524, 492, 460, 428, 396,
> - 372, 356, 340, 324, 308, 292, 276, 260,
> - 244, 228, 212, 196, 180, 164, 148, 132,
> - 120, 112, 104, 96, 88, 80, 72, 64,
> - 56, 48, 40, 32, 24, 16, 8, 0,
> -};
> -
> -/* 16 bit A-law */
> -
> -static short alaw2dma16[] = {
> - -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736,
> - -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784,
> - -2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368,
> - -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392,
> - -22016, -20992, -24064, -23040, -17920, -16896, -19968, -18944,
> - -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136,
> - -11008, -10496, -12032, -11520, -8960, -8448, -9984, -9472,
> - -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568,
> - -344, -328, -376, -360, -280, -264, -312, -296,
> - -472, -456, -504, -488, -408, -392, -440, -424,
> - -88, -72, -120, -104, -24, -8, -56, -40,
> - -216, -200, -248, -232, -152, -136, -184, -168,
> - -1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184,
> - -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696,
> - -688, -656, -752, -720, -560, -528, -624, -592,
> - -944, -912, -1008, -976, -816, -784, -880, -848,
> - 5504, 5248, 6016, 5760, 4480, 4224, 4992, 4736,
> - 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784,
> - 2752, 2624, 3008, 2880, 2240, 2112, 2496, 2368,
> - 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392,
> - 22016, 20992, 24064, 23040, 17920, 16896, 19968, 18944,
> - 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136,
> - 11008, 10496, 12032, 11520, 8960, 8448, 9984, 9472,
> - 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568,
> - 344, 328, 376, 360, 280, 264, 312, 296,
> - 472, 456, 504, 488, 408, 392, 440, 424,
> - 88, 72, 120, 104, 24, 8, 56, 40,
> - 216, 200, 248, 232, 152, 136, 184, 168,
> - 1376, 1312, 1504, 1440, 1120, 1056, 1248, 1184,
> - 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696,
> - 688, 656, 752, 720, 560, 528, 624, 592,
> - 944, 912, 1008, 976, 816, 784, 880, 848,
> -};
> -
> -
> -/*** Translations ************************************************************/
> -
> -
> -static ssize_t cs4218_ct_law(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ct_s8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ct_u8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ct_s16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ct_u16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ctx_law(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ctx_s8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ctx_u8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ctx_s16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ctx_u16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ct_s16_read(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t cs4218_ct_u16_read(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -
> -
> -/*** Low level stuff *********************************************************/
> -
> -struct cs_sound_settings {
> - MACHINE mach; /* machine dependent things */
> - SETTINGS hard; /* hardware settings */
> - SETTINGS soft; /* software settings */
> - SETTINGS dsp; /* /dev/dsp default settings */
> - TRANS *trans_write; /* supported translations for playback */
> - TRANS *trans_read; /* supported translations for record */
> - int volume_left; /* volume (range is machine dependent) */
> - int volume_right;
> - int bass; /* tone (range is machine dependent) */
> - int treble;
> - int gain;
> - int minDev; /* minor device number currently open */
> -};
> -
> -static struct cs_sound_settings sound;
> -
> -static void *CS_Alloc(unsigned int size, gfp_t flags);
> -static void CS_Free(void *ptr, unsigned int size);
> -static int CS_IrqInit(void);
> -#ifdef MODULE
> -static void CS_IrqCleanup(void);
> -#endif /* MODULE */
> -static void CS_Silence(void);
> -static void CS_Init(void);
> -static void CS_Play(void);
> -static void CS_Record(void);
> -static int CS_SetFormat(int format);
> -static int CS_SetVolume(int volume);
> -static void cs4218_tdm_tx_intr(void *devid);
> -static void cs4218_tdm_rx_intr(void *devid);
> -static void cs4218_intr(void *devid);
> -static int cs_get_volume(uint reg);
> -static int cs_volume_setter(int volume, int mute);
> -static int cs_get_gain(uint reg);
> -static int cs_set_gain(int gain);
> -static void cs_mksound(unsigned int hz, unsigned int ticks);
> -static void cs_nosound(unsigned long xx);
> -
> -/*** Mid level stuff *********************************************************/
> -
> -
> -static void sound_silence(void);
> -static void sound_init(void);
> -static int sound_set_format(int format);
> -static int sound_set_speed(int speed);
> -static int sound_set_stereo(int stereo);
> -static int sound_set_volume(int volume);
> -
> -static ssize_t sound_copy_translate(const u_char *userPtr,
> - size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -static ssize_t sound_copy_translate_read(const u_char *userPtr,
> - size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft);
> -
> -
> -/*
> - * /dev/mixer abstraction
> - */
> -
> -struct sound_mixer {
> - int busy;
> - int modify_counter;
> -};
> -
> -static struct sound_mixer mixer;
> -
> -static struct sound_queue sq;
> -static struct sound_queue read_sq;
> -
> -#define sq_block_address(i) (sq.buffers[i])
> -#define SIGNAL_RECEIVED (signal_pending(current))
> -#define NON_BLOCKING(open_mode) (open_mode & O_NONBLOCK)
> -#define ONE_SECOND HZ /* in jiffies (100ths of a second) */
> -#define NO_TIME_LIMIT 0xffffffff
> -
> -/*
> - * /dev/sndstat
> - */
> -
> -struct sound_state {
> - int busy;
> - char buf[512];
> - int len, ptr;
> -};
> -
> -static struct sound_state state;
> -
> -/*** Common stuff ********************************************************/
> -
> -static long long sound_lseek(struct file *file, long long offset, int orig);
> -
> -/*** Config & Setup **********************************************************/
> -
> -void dmasound_setup(char *str, int *ints);
> -
> -/*** Translations ************************************************************/
> -
> -
> -/* ++TeSche: radically changed for new expanding purposes...
> - *
> - * These two routines now deal with copying/expanding/translating the samples
> - * from user space into our buffer at the right frequency. They take care about
> - * how much data there's actually to read, how much buffer space there is and
> - * to convert samples into the right frequency/encoding. They will only work on
> - * complete samples so it may happen they leave some bytes in the input stream
> - * if the user didn't write a multiple of the current sample size. They both
> - * return the number of bytes they've used from both streams so you may detect
> - * such a situation. Luckily all programs should be able to cope with that.
> - *
> - * I think I've optimized anything as far as one can do in plain C, all
> - * variables should fit in registers and the loops are really short. There's
> - * one loop for every possible situation. Writing a more generalized and thus
> - * parameterized loop would only produce slower code. Feel free to optimize
> - * this in assembler if you like. :)
> - *
> - * I think these routines belong here because they're not yet really hardware
> - * independent, especially the fact that the Falcon can play 16bit samples
> - * only in stereo is hardcoded in both of them!
> - *
> - * ++geert: split in even more functions (one per format)
> - */
> -
> -static ssize_t cs4218_ct_law(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - short *table = sound.soft.format == AFMT_MU_LAW ? ulaw2dma16: alaw2dma16;
> - ssize_t count, used;
> - short *p = (short *) &frame[*frameUsed];
> - int val, stereo = sound.soft.stereo;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - u_char data;
> - if (get_user(data, userPtr++))
> - return -EFAULT;
> - val = table[data];
> - *p++ = val;
> - if (stereo) {
> - if (get_user(data, userPtr++))
> - return -EFAULT;
> - val = table[data];
> - }
> - *p++ = val;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 2: used;
> -}
> -
> -
> -static ssize_t cs4218_ct_s8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - short *p = (short *) &frame[*frameUsed];
> - int val, stereo = sound.soft.stereo;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - u_char data;
> - if (get_user(data, userPtr++))
> - return -EFAULT;
> - val = data << 8;
> - *p++ = val;
> - if (stereo) {
> - if (get_user(data, userPtr++))
> - return -EFAULT;
> - val = data << 8;
> - }
> - *p++ = val;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 2: used;
> -}
> -
> -
> -static ssize_t cs4218_ct_u8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - short *p = (short *) &frame[*frameUsed];
> - int val, stereo = sound.soft.stereo;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - u_char data;
> - if (get_user(data, userPtr++))
> - return -EFAULT;
> - val = (data ^ 0x80) << 8;
> - *p++ = val;
> - if (stereo) {
> - if (get_user(data, userPtr++))
> - return -EFAULT;
> - val = (data ^ 0x80) << 8;
> - }
> - *p++ = val;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 2: used;
> -}
> -
> -
> -/* This is the default format of the codec. Signed, 16-bit stereo
> - * generated by an application shouldn't have to be copied at all.
> - * We should just get the phsical address of the buffers and update
> - * the TDM BDs directly.
> - */
> -static ssize_t cs4218_ct_s16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - int stereo = sound.soft.stereo;
> - short *fp = (short *) &frame[*frameUsed];
> -
> - frameLeft >>= 2;
> - userCount >>= (stereo? 2: 1);
> - used = count = min(userCount, frameLeft);
> - if (!stereo) {
> - short *up = (short *) userPtr;
> - while (count > 0) {
> - short data;
> - if (get_user(data, up++))
> - return -EFAULT;
> - *fp++ = data;
> - *fp++ = data;
> - count--;
> - }
> - } else {
> - if (copy_from_user(fp, userPtr, count * 4))
> - return -EFAULT;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 4: used * 2;
> -}
> -
> -static ssize_t cs4218_ct_u16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - int mask = (sound.soft.format == AFMT_U16_LE? 0x0080: 0x8000);
> - int stereo = sound.soft.stereo;
> - short *fp = (short *) &frame[*frameUsed];
> - short *up = (short *) userPtr;
> -
> - frameLeft >>= 2;
> - userCount >>= (stereo? 2: 1);
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - int data;
> - if (get_user(data, up++))
> - return -EFAULT;
> - data ^= mask;
> - *fp++ = data;
> - if (stereo) {
> - if (get_user(data, up++))
> - return -EFAULT;
> - data ^= mask;
> - }
> - *fp++ = data;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 4: used * 2;
> -}
> -
> -
> -static ssize_t cs4218_ctx_law(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - unsigned short *table = (unsigned short *)
> - (sound.soft.format == AFMT_MU_LAW ? ulaw2dma16: alaw2dma16);
> - unsigned int data = expand_data;
> - unsigned int *p = (unsigned int *) &frame[*frameUsed];
> - int bal = expand_bal;
> - int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
> - int utotal, ftotal;
> - int stereo = sound.soft.stereo;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - ftotal = frameLeft;
> - utotal = userCount;
> - while (frameLeft) {
> - u_char c;
> - if (bal < 0) {
> - if (userCount == 0)
> - break;
> - if (get_user(c, userPtr++))
> - return -EFAULT;
> - data = table[c];
> - if (stereo) {
> - if (get_user(c, userPtr++))
> - return -EFAULT;
> - data = (data << 16) + table[c];
> - } else
> - data = (data << 16) + data;
> - userCount--;
> - bal += hSpeed;
> - }
> - *p++ = data;
> - frameLeft--;
> - bal -= sSpeed;
> - }
> - expand_bal = bal;
> - expand_data = data;
> - *frameUsed += (ftotal - frameLeft) * 4;
> - utotal -= userCount;
> - return stereo? utotal * 2: utotal;
> -}
> -
> -
> -static ssize_t cs4218_ctx_s8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - unsigned int *p = (unsigned int *) &frame[*frameUsed];
> - unsigned int data = expand_data;
> - int bal = expand_bal;
> - int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
> - int stereo = sound.soft.stereo;
> - int utotal, ftotal;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - ftotal = frameLeft;
> - utotal = userCount;
> - while (frameLeft) {
> - u_char c;
> - if (bal < 0) {
> - if (userCount == 0)
> - break;
> - if (get_user(c, userPtr++))
> - return -EFAULT;
> - data = c << 8;
> - if (stereo) {
> - if (get_user(c, userPtr++))
> - return -EFAULT;
> - data = (data << 16) + (c << 8);
> - } else
> - data = (data << 16) + data;
> - userCount--;
> - bal += hSpeed;
> - }
> - *p++ = data;
> - frameLeft--;
> - bal -= sSpeed;
> - }
> - expand_bal = bal;
> - expand_data = data;
> - *frameUsed += (ftotal - frameLeft) * 4;
> - utotal -= userCount;
> - return stereo? utotal * 2: utotal;
> -}
> -
> -
> -static ssize_t cs4218_ctx_u8(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - unsigned int *p = (unsigned int *) &frame[*frameUsed];
> - unsigned int data = expand_data;
> - int bal = expand_bal;
> - int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
> - int stereo = sound.soft.stereo;
> - int utotal, ftotal;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - ftotal = frameLeft;
> - utotal = userCount;
> - while (frameLeft) {
> - u_char c;
> - if (bal < 0) {
> - if (userCount == 0)
> - break;
> - if (get_user(c, userPtr++))
> - return -EFAULT;
> - data = (c ^ 0x80) << 8;
> - if (stereo) {
> - if (get_user(c, userPtr++))
> - return -EFAULT;
> - data = (data << 16) + ((c ^ 0x80) << 8);
> - } else
> - data = (data << 16) + data;
> - userCount--;
> - bal += hSpeed;
> - }
> - *p++ = data;
> - frameLeft--;
> - bal -= sSpeed;
> - }
> - expand_bal = bal;
> - expand_data = data;
> - *frameUsed += (ftotal - frameLeft) * 4;
> - utotal -= userCount;
> - return stereo? utotal * 2: utotal;
> -}
> -
> -
> -static ssize_t cs4218_ctx_s16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - unsigned int *p = (unsigned int *) &frame[*frameUsed];
> - unsigned int data = expand_data;
> - unsigned short *up = (unsigned short *) userPtr;
> - int bal = expand_bal;
> - int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
> - int stereo = sound.soft.stereo;
> - int utotal, ftotal;
> -
> - frameLeft >>= 2;
> - userCount >>= (stereo? 2: 1);
> - ftotal = frameLeft;
> - utotal = userCount;
> - while (frameLeft) {
> - unsigned short c;
> - if (bal < 0) {
> - if (userCount == 0)
> - break;
> - if (get_user(data, up++))
> - return -EFAULT;
> - if (stereo) {
> - if (get_user(c, up++))
> - return -EFAULT;
> - data = (data << 16) + c;
> - } else
> - data = (data << 16) + data;
> - userCount--;
> - bal += hSpeed;
> - }
> - *p++ = data;
> - frameLeft--;
> - bal -= sSpeed;
> - }
> - expand_bal = bal;
> - expand_data = data;
> - *frameUsed += (ftotal - frameLeft) * 4;
> - utotal -= userCount;
> - return stereo? utotal * 4: utotal * 2;
> -}
> -
> -
> -static ssize_t cs4218_ctx_u16(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - int mask = (sound.soft.format == AFMT_U16_LE? 0x0080: 0x8000);
> - unsigned int *p = (unsigned int *) &frame[*frameUsed];
> - unsigned int data = expand_data;
> - unsigned short *up = (unsigned short *) userPtr;
> - int bal = expand_bal;
> - int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
> - int stereo = sound.soft.stereo;
> - int utotal, ftotal;
> -
> - frameLeft >>= 2;
> - userCount >>= (stereo? 2: 1);
> - ftotal = frameLeft;
> - utotal = userCount;
> - while (frameLeft) {
> - unsigned short c;
> - if (bal < 0) {
> - if (userCount == 0)
> - break;
> - if (get_user(data, up++))
> - return -EFAULT;
> - data ^= mask;
> - if (stereo) {
> - if (get_user(c, up++))
> - return -EFAULT;
> - data = (data << 16) + (c ^ mask);
> - } else
> - data = (data << 16) + data;
> - userCount--;
> - bal += hSpeed;
> - }
> - *p++ = data;
> - frameLeft--;
> - bal -= sSpeed;
> - }
> - expand_bal = bal;
> - expand_data = data;
> - *frameUsed += (ftotal - frameLeft) * 4;
> - utotal -= userCount;
> - return stereo? utotal * 4: utotal * 2;
> -}
> -
> -static ssize_t cs4218_ct_s8_read(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - short *p = (short *) &frame[*frameUsed];
> - int val, stereo = sound.soft.stereo;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - u_char data;
> -
> - val = *p++;
> - data = val >> 8;
> - if (put_user(data, (u_char *)userPtr++))
> - return -EFAULT;
> - if (stereo) {
> - val = *p;
> - data = val >> 8;
> - if (put_user(data, (u_char *)userPtr++))
> - return -EFAULT;
> - }
> - p++;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 2: used;
> -}
> -
> -
> -static ssize_t cs4218_ct_u8_read(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - short *p = (short *) &frame[*frameUsed];
> - int val, stereo = sound.soft.stereo;
> -
> - frameLeft >>= 2;
> - if (stereo)
> - userCount >>= 1;
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - u_char data;
> -
> - val = *p++;
> - data = (val >> 8) ^ 0x80;
> - if (put_user(data, (u_char *)userPtr++))
> - return -EFAULT;
> - if (stereo) {
> - val = *p;
> - data = (val >> 8) ^ 0x80;
> - if (put_user(data, (u_char *)userPtr++))
> - return -EFAULT;
> - }
> - p++;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 2: used;
> -}
> -
> -
> -static ssize_t cs4218_ct_s16_read(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - int stereo = sound.soft.stereo;
> - short *fp = (short *) &frame[*frameUsed];
> -
> - frameLeft >>= 2;
> - userCount >>= (stereo? 2: 1);
> - used = count = min(userCount, frameLeft);
> - if (!stereo) {
> - short *up = (short *) userPtr;
> - while (count > 0) {
> - short data;
> - data = *fp;
> - if (put_user(data, up++))
> - return -EFAULT;
> - fp+=2;
> - count--;
> - }
> - } else {
> - if (copy_to_user((u_char *)userPtr, fp, count * 4))
> - return -EFAULT;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 4: used * 2;
> -}
> -
> -static ssize_t cs4218_ct_u16_read(const u_char *userPtr, size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t count, used;
> - int mask = (sound.soft.format == AFMT_U16_LE? 0x0080: 0x8000);
> - int stereo = sound.soft.stereo;
> - short *fp = (short *) &frame[*frameUsed];
> - short *up = (short *) userPtr;
> -
> - frameLeft >>= 2;
> - userCount >>= (stereo? 2: 1);
> - used = count = min(userCount, frameLeft);
> - while (count > 0) {
> - int data;
> -
> - data = *fp++;
> - data ^= mask;
> - if (put_user(data, up++))
> - return -EFAULT;
> - if (stereo) {
> - data = *fp;
> - data ^= mask;
> - if (put_user(data, up++))
> - return -EFAULT;
> - }
> - fp++;
> - count--;
> - }
> - *frameUsed += used * 4;
> - return stereo? used * 4: used * 2;
> -}
> -
> -static TRANS transCSNormal = {
> - cs4218_ct_law, cs4218_ct_law, cs4218_ct_s8, cs4218_ct_u8,
> - cs4218_ct_s16, cs4218_ct_u16, cs4218_ct_s16, cs4218_ct_u16
> -};
> -
> -static TRANS transCSExpand = {
> - cs4218_ctx_law, cs4218_ctx_law, cs4218_ctx_s8, cs4218_ctx_u8,
> - cs4218_ctx_s16, cs4218_ctx_u16, cs4218_ctx_s16, cs4218_ctx_u16
> -};
> -
> -static TRANS transCSNormalRead = {
> - NULL, NULL, cs4218_ct_s8_read, cs4218_ct_u8_read,
> - cs4218_ct_s16_read, cs4218_ct_u16_read,
> - cs4218_ct_s16_read, cs4218_ct_u16_read
> -};
> -
> -/*** Low level stuff *********************************************************/
> -
> -static void *CS_Alloc(unsigned int size, gfp_t flags)
> -{
> - int order;
> -
> - size >>= 13;
> - for (order=0; order < 5; order++) {
> - if (size == 0)
> - break;
> - size >>= 1;
> - }
> - return (void *)__get_free_pages(flags, order);
> -}
> -
> -static void CS_Free(void *ptr, unsigned int size)
> -{
> - int order;
> -
> - size >>= 13;
> - for (order=0; order < 5; order++) {
> - if (size == 0)
> - break;
> - size >>= 1;
> - }
> - free_pages((ulong)ptr, order);
> -}
> -
> -static int __init CS_IrqInit(void)
> -{
> - cpm_install_handler(CPMVEC_SMC2, cs4218_intr, NULL);
> - return 1;
> -}
> -
> -#ifdef MODULE
> -static void CS_IrqCleanup(void)
> -{
> - volatile smc_t *sp;
> - volatile cpm8xx_t *cp;
> -
> - /* First disable transmitter and receiver.
> - */
> - sp = &cpmp->cp_smc[1];
> - sp->smc_smcmr &= ~(SMCMR_REN | SMCMR_TEN);
> -
> - /* And now shut down the SMC.
> - */
> - cp = cpmp; /* Get pointer to Communication Processor */
> - cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
> - CPM_CR_STOP_TX) | CPM_CR_FLG;
> - while (cp->cp_cpcr & CPM_CR_FLG);
> -
> - /* Release the interrupt handler.
> - */
> - cpm_free_handler(CPMVEC_SMC2);
> -
> - kfree(beep_buf);
> - kd_mksound = orig_mksound;
> -}
> -#endif /* MODULE */
> -
> -static void CS_Silence(void)
> -{
> - volatile smc_t *sp;
> -
> - /* Disable transmitter.
> - */
> - sp = &cpmp->cp_smc[1];
> - sp->smc_smcmr &= ~SMCMR_TEN;
> -}
> -
> -/* Frequencies depend upon external oscillator. There are two
> - * choices, 12.288 and 11.2896 MHz. The RPCG audio supports both through
> - * and external control register selection bit.
> - */
> -static int cs4218_freqs[] = {
> - /* 12.288 11.2896 */
> - 48000, 44100,
> - 32000, 29400,
> - 24000, 22050,
> - 19200, 17640,
> - 16000, 14700,
> - 12000, 11025,
> - 9600, 8820,
> - 8000, 7350
> -};
> -
> -static void CS_Init(void)
> -{
> - int i, tolerance;
> -
> - switch (sound.soft.format) {
> - case AFMT_S16_LE:
> - case AFMT_U16_LE:
> - sound.hard.format = AFMT_S16_LE;
> - break;
> - default:
> - sound.hard.format = AFMT_S16_BE;
> - break;
> - }
> - sound.hard.stereo = 1;
> - sound.hard.size = 16;
> -
> - /*
> - * If we have a sample rate which is within catchRadius percent
> - * of the requested value, we don't have to expand the samples.
> - * Otherwise choose the next higher rate.
> - */
> - i = (sizeof(cs4218_freqs) / sizeof(int));
> - do {
> - tolerance = catchRadius * cs4218_freqs[--i] / 100;
> - } while (sound.soft.speed > cs4218_freqs[i] + tolerance && i > 0);
> - if (sound.soft.speed >= cs4218_freqs[i] - tolerance)
> - sound.trans_write = &transCSNormal;
> - else
> - sound.trans_write = &transCSExpand;
> - sound.trans_read = &transCSNormalRead;
> - sound.hard.speed = cs4218_freqs[i];
> - cs4218_rate_index = i;
> -
> - /* The CS4218 has seven selectable clock dividers for the sample
> - * clock. The HIOX then provides one of two external rates.
> - * An even numbered frequency table index uses the high external
> - * clock rate.
> - */
> - *(uint *)HIOX_CSR4_ADDR &= ~(HIOX_CSR4_AUDCLKHI | HIOX_CSR4_AUDCLKSEL);
> - if ((i & 1) == 0)
> - *(uint *)HIOX_CSR4_ADDR |= HIOX_CSR4_AUDCLKHI;
> - i >>= 1;
> - *(uint *)HIOX_CSR4_ADDR |= (i & HIOX_CSR4_AUDCLKSEL);
> -
> - expand_bal = -sound.soft.speed;
> -}
> -
> -static int CS_SetFormat(int format)
> -{
> - int size;
> -
> - switch (format) {
> - case AFMT_QUERY:
> - return sound.soft.format;
> - case AFMT_MU_LAW:
> - case AFMT_A_LAW:
> - case AFMT_U8:
> - case AFMT_S8:
> - size = 8;
> - break;
> - case AFMT_S16_BE:
> - case AFMT_U16_BE:
> - case AFMT_S16_LE:
> - case AFMT_U16_LE:
> - size = 16;
> - break;
> - default: /* :-) */
> - printk(KERN_ERR "dmasound: unknown format 0x%x, using AFMT_U8\n",
> - format);
> - size = 8;
> - format = AFMT_U8;
> - }
> -
> - sound.soft.format = format;
> - sound.soft.size = size;
> - if (sound.minDev == SND_DEV_DSP) {
> - sound.dsp.format = format;
> - sound.dsp.size = size;
> - }
> -
> - CS_Init();
> -
> - return format;
> -}
> -
> -/* Volume is the amount of attenuation we tell the codec to impose
> - * on the outputs. There are 32 levels, with 0 the "loudest".
> - */
> -#define CS_VOLUME_TO_MASK(x) (31 - ((((x) - 1) * 31) / 99))
> -#define CS_MASK_TO_VOLUME(y) (100 - ((y) * 99 / 31))
> -
> -static int cs_get_volume(uint reg)
> -{
> - int volume;
> -
> - volume = CS_MASK_TO_VOLUME(CS_LATTEN_GET(reg));
> - volume |= CS_MASK_TO_VOLUME(CS_RATTEN_GET(reg)) << 8;
> - return volume;
> -}
> -
> -static int cs_volume_setter(int volume, int mute)
> -{
> - uint tempctl;
> -
> - if (mute && volume == 0) {
> - tempctl = cs4218_control | CS_MUTE;
> - } else {
> - tempctl = cs4218_control & ~CS_MUTE;
> - tempctl = tempctl & ~(CS_LATTEN | CS_RATTEN);
> - tempctl |= CS_LATTEN_SET(CS_VOLUME_TO_MASK(volume & 0xff));
> - tempctl |= CS_RATTEN_SET(CS_VOLUME_TO_MASK((volume >> 8) & 0xff));
> - volume = cs_get_volume(tempctl);
> - }
> - if (tempctl != cs4218_control) {
> - cs4218_ctl_write(tempctl);
> - }
> - return volume;
> -}
> -
> -
> -/* Gain has 16 steps from 0 to 15. These are in 1.5dB increments from
> - * 0 (no gain) to 22.5 dB.
> - */
> -#define CS_RECLEVEL_TO_GAIN(v) \
> - ((v) < 0 ? 0 : (v) > 100 ? 15 : (v) * 3 / 20)
> -#define CS_GAIN_TO_RECLEVEL(v) (((v) * 20 + 2) / 3)
> -
> -static int cs_get_gain(uint reg)
> -{
> - int gain;
> -
> - gain = CS_GAIN_TO_RECLEVEL(CS_LGAIN_GET(reg));
> - gain |= CS_GAIN_TO_RECLEVEL(CS_RGAIN_GET(reg)) << 8;
> - return gain;
> -}
> -
> -static int cs_set_gain(int gain)
> -{
> - uint tempctl;
> -
> - tempctl = cs4218_control & ~(CS_LGAIN | CS_RGAIN);
> - tempctl |= CS_LGAIN_SET(CS_RECLEVEL_TO_GAIN(gain & 0xff));
> - tempctl |= CS_RGAIN_SET(CS_RECLEVEL_TO_GAIN((gain >> 8) & 0xff));
> - gain = cs_get_gain(tempctl);
> -
> - if (tempctl != cs4218_control) {
> - cs4218_ctl_write(tempctl);
> - }
> - return gain;
> -}
> -
> -static int CS_SetVolume(int volume)
> -{
> - return cs_volume_setter(volume, CS_MUTE);
> -}
> -
> -static void CS_Play(void)
> -{
> - int i, count;
> - unsigned long flags;
> - volatile cbd_t *bdp;
> - volatile cpm8xx_t *cp;
> -
> - /* Protect buffer */
> - spin_lock_irqsave(&cs4218_lock, flags);
> -#if 0
> - if (awacs_beep_state) {
> - /* sound takes precedence over beeps */
> - out_le32(&awacs_txdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16);
> - out_le32(&awacs->control,
> - (in_le32(&awacs->control) & ~0x1f00)
> - | (awacs_rate_index << 8));
> - out_le32(&awacs->byteswap, sound.hard.format != AFMT_S16_BE);
> - out_le32(&awacs_txdma->cmdptr, virt_to_bus(&(awacs_tx_cmds[(sq.front+sq.active) % sq.max_count])));
> -
> - beep_playing = 0;
> - awacs_beep_state = 0;
> - }
> -#endif
> - i = sq.front + sq.active;
> - if (i >= sq.max_count)
> - i -= sq.max_count;
> - while (sq.active < 2 && sq.active < sq.count) {
> - count = (sq.count == sq.active + 1)?sq.rear_size:sq.block_size;
> - if (count < sq.block_size && !sq.syncing)
> - /* last block not yet filled, and we're not syncing. */
> - break;
> -
> - bdp = &tx_base[i];
> - bdp->cbd_datlen = count;
> -
> - flush_dcache_range((ulong)sound_buffers[i],
> - (ulong)(sound_buffers[i] + count));
> -
> - if (++i >= sq.max_count)
> - i = 0;
> -
> - if (sq.active == 0) {
> - /* The SMC does not load its fifo until the first
> - * TDM frame pulse, so the transmit data gets shifted
> - * by one word. To compensate for this, we incorrectly
> - * transmit the first buffer and shorten it by one
> - * word. Subsequent buffers are then aligned properly.
> - */
> - bdp->cbd_datlen -= 2;
> -
> - /* Start up the SMC Transmitter.
> - */
> - cp = cpmp;
> - cp->cp_smc[1].smc_smcmr |= SMCMR_TEN;
> - cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
> - CPM_CR_RESTART_TX) | CPM_CR_FLG;
> - while (cp->cp_cpcr & CPM_CR_FLG);
> - }
> -
> - /* Buffer is ready now.
> - */
> - bdp->cbd_sc |= BD_SC_READY;
> -
> - ++sq.active;
> - }
> - spin_unlock_irqrestore(&cs4218_lock, flags);
> -}
> -
> -
> -static void CS_Record(void)
> -{
> - unsigned long flags;
> - volatile smc_t *sp;
> -
> - if (read_sq.active)
> - return;
> -
> - /* Protect buffer */
> - spin_lock_irqsave(&cs4218_lock, flags);
> -
> - /* This is all we have to do......Just start it up.
> - */
> - sp = &cpmp->cp_smc[1];
> - sp->smc_smcmr |= SMCMR_REN;
> -
> - read_sq.active = 1;
> -
> - spin_unlock_irqrestore(&cs4218_lock, flags);
> -}
> -
> -
> -static void
> -cs4218_tdm_tx_intr(void *devid)
> -{
> - int i = sq.front;
> - volatile cbd_t *bdp;
> -
> - while (sq.active > 0) {
> - bdp = &tx_base[i];
> - if (bdp->cbd_sc & BD_SC_READY)
> - break; /* this frame is still going */
> - --sq.count;
> - --sq.active;
> - if (++i >= sq.max_count)
> - i = 0;
> - }
> - if (i != sq.front)
> - WAKE_UP(sq.action_queue);
> - sq.front = i;
> -
> - CS_Play();
> -
> - if (!sq.active)
> - WAKE_UP(sq.sync_queue);
> -}
> -
> -
> -static void
> -cs4218_tdm_rx_intr(void *devid)
> -{
> -
> - /* We want to blow 'em off when shutting down.
> - */
> - if (read_sq.active == 0)
> - return;
> -
> - /* Check multiple buffers in case we were held off from
> - * interrupt processing for a long time. Geeze, I really hope
> - * this doesn't happen.
> - */
> - while ((rx_base[read_sq.rear].cbd_sc & BD_SC_EMPTY) == 0) {
> -
> - /* Invalidate the data cache range for this buffer.
> - */
> - invalidate_dcache_range(
> - (uint)(sound_read_buffers[read_sq.rear]),
> - (uint)(sound_read_buffers[read_sq.rear] + read_sq.block_size));
> -
> - /* Make buffer available again and move on.
> - */
> - rx_base[read_sq.rear].cbd_sc |= BD_SC_EMPTY;
> - read_sq.rear++;
> -
> - /* Wrap the buffer ring.
> - */
> - if (read_sq.rear >= read_sq.max_active)
> - read_sq.rear = 0;
> -
> - /* If we have caught up to the front buffer, bump it.
> - * This will cause weird (but not fatal) results if the
> - * read loop is currently using this buffer. The user is
> - * behind in this case anyway, so weird things are going
> - * to happen.
> - */
> - if (read_sq.rear == read_sq.front) {
> - read_sq.front++;
> - if (read_sq.front >= read_sq.max_active)
> - read_sq.front = 0;
> - }
> - }
> -
> - WAKE_UP(read_sq.action_queue);
> -}
> -
> -static void cs_nosound(unsigned long xx)
> -{
> - unsigned long flags;
> -
> - /* not sure if this is needed, since hardware command is #if 0'd */
> - spin_lock_irqsave(&cs4218_lock, flags);
> - if (beep_playing) {
> -#if 0
> - st_le16(&beep_dbdma_cmd->command, DBDMA_STOP);
> -#endif
> - beep_playing = 0;
> - }
> - spin_unlock_irqrestore(&cs4218_lock, flags);
> -}
> -
> -static DEFINE_TIMER(beep_timer, cs_nosound, 0, 0);
> -
> -static void cs_mksound(unsigned int hz, unsigned int ticks)
> -{
> - unsigned long flags;
> - int beep_speed = BEEP_SPEED;
> - int srate = cs4218_freqs[beep_speed];
> - int period, ncycles, nsamples;
> - int i, j, f;
> - short *p;
> - static int beep_hz_cache;
> - static int beep_nsamples_cache;
> - static int beep_volume_cache;
> -
> - if (hz <= srate / BEEP_BUFLEN || hz > srate / 2) {
> -#if 1
> - /* this is a hack for broken X server code */
> - hz = 750;
> - ticks = 12;
> -#else
> - /* cancel beep currently playing */
> - awacs_nosound(0);
> - return;
> -#endif
> - }
> - /* lock while modifying beep_timer */
> - spin_lock_irqsave(&cs4218_lock, flags);
> - del_timer(&beep_timer);
> - if (ticks) {
> - beep_timer.expires = jiffies + ticks;
> - add_timer(&beep_timer);
> - }
> - if (beep_playing || sq.active || beep_buf == NULL) {
> - spin_unlock_irqrestore(&cs4218_lock, flags);
> - return; /* too hard, sorry :-( */
> - }
> - beep_playing = 1;
> -#if 0
> - st_le16(&beep_dbdma_cmd->command, OUTPUT_MORE + BR_ALWAYS);
> -#endif
> - spin_unlock_irqrestore(&cs4218_lock, flags);
> -
> - if (hz == beep_hz_cache && beep_volume == beep_volume_cache) {
> - nsamples = beep_nsamples_cache;
> - } else {
> - period = srate * 256 / hz; /* fixed point */
> - ncycles = BEEP_BUFLEN * 256 / period;
> - nsamples = (period * ncycles) >> 8;
> - f = ncycles * 65536 / nsamples;
> - j = 0;
> - p = beep_buf;
> - for (i = 0; i < nsamples; ++i, p += 2) {
> - p[0] = p[1] = beep_wform[j >> 8] * beep_volume;
> - j = (j + f) & 0xffff;
> - }
> - beep_hz_cache = hz;
> - beep_volume_cache = beep_volume;
> - beep_nsamples_cache = nsamples;
> - }
> -
> -#if 0
> - st_le16(&beep_dbdma_cmd->req_count, nsamples*4);
> - st_le16(&beep_dbdma_cmd->xfer_status, 0);
> - st_le32(&beep_dbdma_cmd->cmd_dep, virt_to_bus(beep_dbdma_cmd));
> - st_le32(&beep_dbdma_cmd->phy_addr, virt_to_bus(beep_buf));
> - awacs_beep_state = 1;
> -
> - spin_lock_irqsave(&cs4218_lock, flags);
> - if (beep_playing) { /* i.e. haven't been terminated already */
> - out_le32(&awacs_txdma->control, (RUN|WAKE|FLUSH|PAUSE) << 16);
> - out_le32(&awacs->control,
> - (in_le32(&awacs->control) & ~0x1f00)
> - | (beep_speed << 8));
> - out_le32(&awacs->byteswap, 0);
> - out_le32(&awacs_txdma->cmdptr, virt_to_bus(beep_dbdma_cmd));
> - out_le32(&awacs_txdma->control, RUN | (RUN << 16));
> - }
> - spin_unlock_irqrestore(&cs4218_lock, flags);
> -#endif
> -}
> -
> -static MACHINE mach_cs4218 = {
> - .owner = THIS_MODULE,
> - .name = "HIOX CS4218",
> - .name2 = "Built-in Sound",
> - .dma_alloc = CS_Alloc,
> - .dma_free = CS_Free,
> - .irqinit = CS_IrqInit,
> -#ifdef MODULE
> - .irqcleanup = CS_IrqCleanup,
> -#endif /* MODULE */
> - .init = CS_Init,
> - .silence = CS_Silence,
> - .setFormat = CS_SetFormat,
> - .setVolume = CS_SetVolume,
> - .play = CS_Play
> -};
> -
> -
> -/*** Mid level stuff *********************************************************/
> -
> -
> -static void sound_silence(void)
> -{
> - /* update hardware settings one more */
> - (*sound.mach.init)();
> -
> - (*sound.mach.silence)();
> -}
> -
> -
> -static void sound_init(void)
> -{
> - (*sound.mach.init)();
> -}
> -
> -
> -static int sound_set_format(int format)
> -{
> - return(*sound.mach.setFormat)(format);
> -}
> -
> -
> -static int sound_set_speed(int speed)
> -{
> - if (speed < 0)
> - return(sound.soft.speed);
> -
> - sound.soft.speed = speed;
> - (*sound.mach.init)();
> - if (sound.minDev == SND_DEV_DSP)
> - sound.dsp.speed = sound.soft.speed;
> -
> - return(sound.soft.speed);
> -}
> -
> -
> -static int sound_set_stereo(int stereo)
> -{
> - if (stereo < 0)
> - return(sound.soft.stereo);
> -
> - stereo = !!stereo; /* should be 0 or 1 now */
> -
> - sound.soft.stereo = stereo;
> - if (sound.minDev == SND_DEV_DSP)
> - sound.dsp.stereo = stereo;
> - (*sound.mach.init)();
> -
> - return(stereo);
> -}
> -
> -
> -static int sound_set_volume(int volume)
> -{
> - return(*sound.mach.setVolume)(volume);
> -}
> -
> -static ssize_t sound_copy_translate(const u_char *userPtr,
> - size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t (*ct_func)(const u_char *, size_t, u_char *, ssize_t *, ssize_t) = NULL;
> -
> - switch (sound.soft.format) {
> - case AFMT_MU_LAW:
> - ct_func = sound.trans_write->ct_ulaw;
> - break;
> - case AFMT_A_LAW:
> - ct_func = sound.trans_write->ct_alaw;
> - break;
> - case AFMT_S8:
> - ct_func = sound.trans_write->ct_s8;
> - break;
> - case AFMT_U8:
> - ct_func = sound.trans_write->ct_u8;
> - break;
> - case AFMT_S16_BE:
> - ct_func = sound.trans_write->ct_s16be;
> - break;
> - case AFMT_U16_BE:
> - ct_func = sound.trans_write->ct_u16be;
> - break;
> - case AFMT_S16_LE:
> - ct_func = sound.trans_write->ct_s16le;
> - break;
> - case AFMT_U16_LE:
> - ct_func = sound.trans_write->ct_u16le;
> - break;
> - }
> - if (ct_func)
> - return ct_func(userPtr, userCount, frame, frameUsed, frameLeft);
> - else
> - return 0;
> -}
> -
> -static ssize_t sound_copy_translate_read(const u_char *userPtr,
> - size_t userCount,
> - u_char frame[], ssize_t *frameUsed,
> - ssize_t frameLeft)
> -{
> - ssize_t (*ct_func)(const u_char *, size_t, u_char *, ssize_t *, ssize_t) = NULL;
> -
> - switch (sound.soft.format) {
> - case AFMT_MU_LAW:
> - ct_func = sound.trans_read->ct_ulaw;
> - break;
> - case AFMT_A_LAW:
> - ct_func = sound.trans_read->ct_alaw;
> - break;
> - case AFMT_S8:
> - ct_func = sound.trans_read->ct_s8;
> - break;
> - case AFMT_U8:
> - ct_func = sound.trans_read->ct_u8;
> - break;
> - case AFMT_S16_BE:
> - ct_func = sound.trans_read->ct_s16be;
> - break;
> - case AFMT_U16_BE:
> - ct_func = sound.trans_read->ct_u16be;
> - break;
> - case AFMT_S16_LE:
> - ct_func = sound.trans_read->ct_s16le;
> - break;
> - case AFMT_U16_LE:
> - ct_func = sound.trans_read->ct_u16le;
> - break;
> - }
> - if (ct_func)
> - return ct_func(userPtr, userCount, frame, frameUsed, frameLeft);
> - else
> - return 0;
> -}
> -
> -
> -/*
> - * /dev/mixer abstraction
> - */
> -
> -static int mixer_open(struct inode *inode, struct file *file)
> -{
> - mixer.busy = 1;
> - return nonseekable_open(inode, file);
> -}
> -
> -
> -static int mixer_release(struct inode *inode, struct file *file)
> -{
> - mixer.busy = 0;
> - return 0;
> -}
> -
> -
> -static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd,
> - u_long arg)
> -{
> - int data;
> - uint tmpcs;
> -
> - if (_SIOC_DIR(cmd) & _SIOC_WRITE)
> - mixer.modify_counter++;
> - if (cmd == OSS_GETVERSION)
> - return IOCTL_OUT(arg, SOUND_VERSION);
> - switch (cmd) {
> - case SOUND_MIXER_INFO: {
> - mixer_info info;
> - strlcpy(info.id, "CS4218_TDM", sizeof(info.id));
> - strlcpy(info.name, "CS4218_TDM", sizeof(info.name));
> - info.name[sizeof(info.name)-1] = 0;
> - info.modify_counter = mixer.modify_counter;
> - if (copy_to_user((int *)arg, &info, sizeof(info)))
> - return -EFAULT;
> - return 0;
> - }
> - case SOUND_MIXER_READ_DEVMASK:
> - data = SOUND_MASK_VOLUME | SOUND_MASK_LINE
> - | SOUND_MASK_MIC | SOUND_MASK_RECLEV
> - | SOUND_MASK_ALTPCM;
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_READ_RECMASK:
> - data = SOUND_MASK_LINE | SOUND_MASK_MIC;
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_READ_RECSRC:
> - if (cs4218_control & CS_DO1)
> - data = SOUND_MASK_LINE;
> - else
> - data = SOUND_MASK_MIC;
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_WRITE_RECSRC:
> - IOCTL_IN(arg, data);
> - data &= (SOUND_MASK_LINE | SOUND_MASK_MIC);
> - if (data & SOUND_MASK_LINE)
> - tmpcs = cs4218_control |
> - (CS_ISL | CS_ISR | CS_DO1);
> - if (data & SOUND_MASK_MIC)
> - tmpcs = cs4218_control &
> - ~(CS_ISL | CS_ISR | CS_DO1);
> - if (tmpcs != cs4218_control)
> - cs4218_ctl_write(tmpcs);
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_READ_STEREODEVS:
> - data = SOUND_MASK_VOLUME | SOUND_MASK_RECLEV;
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_READ_CAPS:
> - return IOCTL_OUT(arg, 0);
> - case SOUND_MIXER_READ_VOLUME:
> - data = (cs4218_control & CS_MUTE)? 0:
> - cs_get_volume(cs4218_control);
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_WRITE_VOLUME:
> - IOCTL_IN(arg, data);
> - return IOCTL_OUT(arg, sound_set_volume(data));
> - case SOUND_MIXER_WRITE_ALTPCM: /* really bell volume */
> - IOCTL_IN(arg, data);
> - beep_volume = data & 0xff;
> - /* fall through */
> - case SOUND_MIXER_READ_ALTPCM:
> - return IOCTL_OUT(arg, beep_volume);
> - case SOUND_MIXER_WRITE_RECLEV:
> - IOCTL_IN(arg, data);
> - data = cs_set_gain(data);
> - return IOCTL_OUT(arg, data);
> - case SOUND_MIXER_READ_RECLEV:
> - data = cs_get_gain(cs4218_control);
> - return IOCTL_OUT(arg, data);
> - }
> -
> - return -EINVAL;
> -}
> -
> -
> -static const struct file_operations mixer_fops =
> -{
> - .owner = THIS_MODULE,
> - .llseek = sound_lseek,
> - .ioctl = mixer_ioctl,
> - .open = mixer_open,
> - .release = mixer_release,
> -};
> -
> -
> -static void __init mixer_init(void)
> -{
> - mixer_unit = register_sound_mixer(&mixer_fops, -1);
> - if (mixer_unit < 0)
> - return;
> -
> - mixer.busy = 0;
> - sound.treble = 0;
> - sound.bass = 0;
> -
> - /* Set Line input, no gain, no attenuation.
> - */
> - cs4218_control = CS_ISL | CS_ISR | CS_DO1;
> - cs4218_control |= CS_LGAIN_SET(0) | CS_RGAIN_SET(0);
> - cs4218_control |= CS_LATTEN_SET(0) | CS_RATTEN_SET(0);
> - cs4218_ctl_write(cs4218_control);
> -}
> -
> -
> -/*
> - * Sound queue stuff, the heart of the driver
> - */
> -
> -
> -static int sq_allocate_buffers(void)
> -{
> - int i;
> -
> - if (sound_buffers)
> - return 0;
> - sound_buffers = kmalloc (numBufs * sizeof(char *), GFP_KERNEL);
> - if (!sound_buffers)
> - return -ENOMEM;
> - for (i = 0; i < numBufs; i++) {
> - sound_buffers[i] = sound.mach.dma_alloc (bufSize << 10, GFP_KERNEL);
> - if (!sound_buffers[i]) {
> - while (i--)
> - sound.mach.dma_free (sound_buffers[i], bufSize << 10);
> - kfree (sound_buffers);
> - sound_buffers = 0;
> - return -ENOMEM;
> - }
> - }
> - return 0;
> -}
> -
> -
> -static void sq_release_buffers(void)
> -{
> - int i;
> -
> - if (sound_buffers) {
> - for (i = 0; i < numBufs; i++)
> - sound.mach.dma_free (sound_buffers[i], bufSize << 10);
> - kfree (sound_buffers);
> - sound_buffers = 0;
> - }
> -}
> -
> -
> -static int sq_allocate_read_buffers(void)
> -{
> - int i;
> -
> - if (sound_read_buffers)
> - return 0;
> - sound_read_buffers = kmalloc(numReadBufs * sizeof(char *), GFP_KERNEL);
> - if (!sound_read_buffers)
> - return -ENOMEM;
> - for (i = 0; i < numBufs; i++) {
> - sound_read_buffers[i] = sound.mach.dma_alloc (readbufSize<<10,
> - GFP_KERNEL);
> - if (!sound_read_buffers[i]) {
> - while (i--)
> - sound.mach.dma_free (sound_read_buffers[i],
> - readbufSize << 10);
> - kfree (sound_read_buffers);
> - sound_read_buffers = 0;
> - return -ENOMEM;
> - }
> - }
> - return 0;
> -}
> -
> -static void sq_release_read_buffers(void)
> -{
> - int i;
> -
> - if (sound_read_buffers) {
> - cpmp->cp_smc[1].smc_smcmr &= ~SMCMR_REN;
> - for (i = 0; i < numReadBufs; i++)
> - sound.mach.dma_free (sound_read_buffers[i],
> - bufSize << 10);
> - kfree (sound_read_buffers);
> - sound_read_buffers = 0;
> - }
> -}
> -
> -
> -static void sq_setup(int numBufs, int bufSize, char **write_buffers)
> -{
> - int i;
> - volatile cbd_t *bdp;
> - volatile cpm8xx_t *cp;
> - volatile smc_t *sp;
> -
> - /* Make sure the SMC transmit is shut down.
> - */
> - cp = cpmp;
> - sp = &cpmp->cp_smc[1];
> - sp->smc_smcmr &= ~SMCMR_TEN;
> -
> - sq.max_count = numBufs;
> - sq.max_active = numBufs;
> - sq.block_size = bufSize;
> - sq.buffers = write_buffers;
> -
> - sq.front = sq.count = 0;
> - sq.rear = -1;
> - sq.syncing = 0;
> - sq.active = 0;
> -
> - bdp = tx_base;
> - for (i=0; i<numBufs; i++) {
> - bdp->cbd_bufaddr = virt_to_bus(write_buffers[i]);
> - bdp++;
> - }
> -
> - /* This causes the SMC to sync up with the first buffer again.
> - */
> - cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2, CPM_CR_INIT_TX) | CPM_CR_FLG;
> - while (cp->cp_cpcr & CPM_CR_FLG);
> -}
> -
> -static void read_sq_setup(int numBufs, int bufSize, char **read_buffers)
> -{
> - int i;
> - volatile cbd_t *bdp;
> - volatile cpm8xx_t *cp;
> - volatile smc_t *sp;
> -
> - /* Make sure the SMC receive is shut down.
> - */
> - cp = cpmp;
> - sp = &cpmp->cp_smc[1];
> - sp->smc_smcmr &= ~SMCMR_REN;
> -
> - read_sq.max_count = numBufs;
> - read_sq.max_active = numBufs;
> - read_sq.block_size = bufSize;
> - read_sq.buffers = read_buffers;
> -
> - read_sq.front = read_sq.count = 0;
> - read_sq.rear = 0;
> - read_sq.rear_size = 0;
> - read_sq.syncing = 0;
> - read_sq.active = 0;
> -
> - bdp = rx_base;
> - for (i=0; i<numReadBufs; i++) {
> - bdp->cbd_bufaddr = virt_to_bus(read_buffers[i]);
> - bdp->cbd_datlen = read_sq.block_size;
> - bdp++;
> - }
> -
> - /* This causes the SMC to sync up with the first buffer again.
> - */
> - cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2, CPM_CR_INIT_RX) | CPM_CR_FLG;
> - while (cp->cp_cpcr & CPM_CR_FLG);
> -}
> -
> -
> -static void sq_play(void)
> -{
> - (*sound.mach.play)();
> -}
> -
> -
> -/* ++TeSche: radically changed this one too */
> -
> -static ssize_t sq_write(struct file *file, const char *src, size_t uLeft,
> - loff_t *ppos)
> -{
> - ssize_t uWritten = 0;
> - u_char *dest;
> - ssize_t uUsed, bUsed, bLeft;
> -
> - /* ++TeSche: Is something like this necessary?
> - * Hey, that's an honest question! Or does any other part of the
> - * filesystem already checks this situation? I really don't know.
> - */
> - if (uLeft == 0)
> - return 0;
> -
> - /* The interrupt doesn't start to play the last, incomplete frame.
> - * Thus we can append to it without disabling the interrupts! (Note
> - * also that sq.rear isn't affected by the interrupt.)
> - */
> -
> - if (sq.count > 0 && (bLeft = sq.block_size-sq.rear_size) > 0) {
> - dest = sq_block_address(sq.rear);
> - bUsed = sq.rear_size;
> - uUsed = sound_copy_translate(src, uLeft, dest, &bUsed, bLeft);
> - if (uUsed <= 0)
> - return uUsed;
> - src += uUsed;
> - uWritten += uUsed;
> - uLeft -= uUsed;
> - sq.rear_size = bUsed;
> - }
> -
> - do {
> - while (sq.count == sq.max_active) {
> - sq_play();
> - if (NON_BLOCKING(sq.open_mode))
> - return uWritten > 0 ? uWritten : -EAGAIN;
> - SLEEP(sq.action_queue);
> - if (SIGNAL_RECEIVED)
> - return uWritten > 0 ? uWritten : -EINTR;
> - }
> -
> - /* Here, we can avoid disabling the interrupt by first
> - * copying and translating the data, and then updating
> - * the sq variables. Until this is done, the interrupt
> - * won't see the new frame and we can work on it
> - * undisturbed.
> - */
> -
> - dest = sq_block_address((sq.rear+1) % sq.max_count);
> - bUsed = 0;
> - bLeft = sq.block_size;
> - uUsed = sound_copy_translate(src, uLeft, dest, &bUsed, bLeft);
> - if (uUsed <= 0)
> - break;
> - src += uUsed;
> - uWritten += uUsed;
> - uLeft -= uUsed;
> - if (bUsed) {
> - sq.rear = (sq.rear+1) % sq.max_count;
> - sq.rear_size = bUsed;
> - sq.count++;
> - }
> - } while (bUsed); /* uUsed may have been 0 */
> -
> - sq_play();
> -
> - return uUsed < 0? uUsed: uWritten;
> -}
> -
> -
> -/***********/
> -
> -/* Here is how the values are used for reading.
> - * The value 'active' simply indicates the DMA is running. This is
> - * done so the driver semantics are DMA starts when the first read is
> - * posted. The value 'front' indicates the buffer we should next
> - * send to the user. The value 'rear' indicates the buffer the DMA is
> - * currently filling. When 'front' == 'rear' the buffer "ring" is
> - * empty (we always have an empty available). The 'rear_size' is used
> - * to track partial offsets into the current buffer. Right now, I just keep
> - * The DMA running. If the reader can't keep up, the interrupt tosses
> - * the oldest buffer. We could also shut down the DMA in this case.
> - */
> -static ssize_t sq_read(struct file *file, char *dst, size_t uLeft,
> - loff_t *ppos)
> -{
> -
> - ssize_t uRead, bLeft, bUsed, uUsed;
> -
> - if (uLeft == 0)
> - return 0;
> -
> - if (!read_sq.active)
> - CS_Record(); /* Kick off the record process. */
> -
> - uRead = 0;
> -
> - /* Move what the user requests, depending upon other options.
> - */
> - while (uLeft > 0) {
> -
> - /* When front == rear, the DMA is not done yet.
> - */
> - while (read_sq.front == read_sq.rear) {
> - if (NON_BLOCKING(read_sq.open_mode)) {
> - return uRead > 0 ? uRead : -EAGAIN;
> - }
> - SLEEP(read_sq.action_queue);
> - if (SIGNAL_RECEIVED)
> - return uRead > 0 ? uRead : -EINTR;
> - }
> -
> - /* The amount we move is either what is left in the
> - * current buffer or what the user wants.
> - */
> - bLeft = read_sq.block_size - read_sq.rear_size;
> - bUsed = read_sq.rear_size;
> - uUsed = sound_copy_translate_read(dst, uLeft,
> - read_sq.buffers[read_sq.front], &bUsed, bLeft);
> - if (uUsed <= 0)
> - return uUsed;
> - dst += uUsed;
> - uRead += uUsed;
> - uLeft -= uUsed;
> - read_sq.rear_size += bUsed;
> - if (read_sq.rear_size >= read_sq.block_size) {
> - read_sq.rear_size = 0;
> - read_sq.front++;
> - if (read_sq.front >= read_sq.max_active)
> - read_sq.front = 0;
> - }
> - }
> - return uRead;
> -}
> -
> -static int sq_open(struct inode *inode, struct file *file)
> -{
> - int rc = 0;
> -
> - if (file->f_mode & FMODE_WRITE) {
> - if (sq.busy) {
> - rc = -EBUSY;
> - if (NON_BLOCKING(file->f_flags))
> - goto err_out;
> - rc = -EINTR;
> - while (sq.busy) {
> - SLEEP(sq.open_queue);
> - if (SIGNAL_RECEIVED)
> - goto err_out;
> - }
> - }
> - sq.busy = 1; /* Let's play spot-the-race-condition */
> -
> - if (sq_allocate_buffers()) goto err_out_nobusy;
> -
> - sq_setup(numBufs, bufSize<<10,sound_buffers);
> - sq.open_mode = file->f_mode;
> - }
> -
> -
> - if (file->f_mode & FMODE_READ) {
> - if (read_sq.busy) {
> - rc = -EBUSY;
> - if (NON_BLOCKING(file->f_flags))
> - goto err_out;
> - rc = -EINTR;
> - while (read_sq.busy) {
> - SLEEP(read_sq.open_queue);
> - if (SIGNAL_RECEIVED)
> - goto err_out;
> - }
> - rc = 0;
> - }
> - read_sq.busy = 1;
> - if (sq_allocate_read_buffers()) goto err_out_nobusy;
> -
> - read_sq_setup(numReadBufs,readbufSize<<10, sound_read_buffers);
> - read_sq.open_mode = file->f_mode;
> - }
> -
> - /* Start up the 4218 by:
> - * Reset.
> - * Enable, unreset.
> - */
> - *((volatile uint *)HIOX_CSR4_ADDR) &= ~HIOX_CSR4_RSTAUDIO;
> - eieio();
> - *((volatile uint *)HIOX_CSR4_ADDR) |= HIOX_CSR4_ENAUDIO;
> - mdelay(50);
> - *((volatile uint *)HIOX_CSR4_ADDR) |= HIOX_CSR4_RSTAUDIO;
> -
> - /* We need to send the current control word in case someone
> - * opened /dev/mixer and changed things while we were shut
> - * down. Chances are good the initialization that follows
> - * would have done this, but it is still possible it wouldn't.
> - */
> - cs4218_ctl_write(cs4218_control);
> -
> - sound.minDev = iminor(inode) & 0x0f;
> - sound.soft = sound.dsp;
> - sound.hard = sound.dsp;
> - sound_init();
> - if ((iminor(inode) & 0x0f) == SND_DEV_AUDIO) {
> - sound_set_speed(8000);
> - sound_set_stereo(0);
> - sound_set_format(AFMT_MU_LAW);
> - }
> -
> - return nonseekable_open(inode, file);
> -
> -err_out_nobusy:
> - if (file->f_mode & FMODE_WRITE) {
> - sq.busy = 0;
> - WAKE_UP(sq.open_queue);
> - }
> - if (file->f_mode & FMODE_READ) {
> - read_sq.busy = 0;
> - WAKE_UP(read_sq.open_queue);
> - }
> -err_out:
> - return rc;
> -}
> -
> -
> -static void sq_reset(void)
> -{
> - sound_silence();
> - sq.active = 0;
> - sq.count = 0;
> - sq.front = (sq.rear+1) % sq.max_count;
> -#if 0
> - init_tdm_buffers();
> -#endif
> -}
> -
> -
> -static int sq_fsync(struct file *filp, struct dentry *dentry)
> -{
> - int rc = 0;
> -
> - sq.syncing = 1;
> - sq_play(); /* there may be an incomplete frame waiting */
> -
> - while (sq.active) {
> - SLEEP(sq.sync_queue);
> - if (SIGNAL_RECEIVED) {
> - /* While waiting for audio output to drain, an
> - * interrupt occurred. Stop audio output immediately
> - * and clear the queue. */
> - sq_reset();
> - rc = -EINTR;
> - break;
> - }
> - }
> -
> - sq.syncing = 0;
> - return rc;
> -}
> -
> -static int sq_release(struct inode *inode, struct file *file)
> -{
> - int rc = 0;
> -
> - if (sq.busy)
> - rc = sq_fsync(file, file->f_path.dentry);
> - sound.soft = sound.dsp;
> - sound.hard = sound.dsp;
> - sound_silence();
> -
> - sq_release_read_buffers();
> - sq_release_buffers();
> -
> - if (file->f_mode & FMODE_READ) {
> - read_sq.busy = 0;
> - WAKE_UP(read_sq.open_queue);
> - }
> -
> - if (file->f_mode & FMODE_WRITE) {
> - sq.busy = 0;
> - WAKE_UP(sq.open_queue);
> - }
> -
> - /* Shut down the SMC.
> - */
> - cpmp->cp_smc[1].smc_smcmr &= ~(SMCMR_TEN | SMCMR_REN);
> -
> - /* Shut down the codec.
> - */
> - *((volatile uint *)HIOX_CSR4_ADDR) |= HIOX_CSR4_RSTAUDIO;
> - eieio();
> - *((volatile uint *)HIOX_CSR4_ADDR) &= ~HIOX_CSR4_ENAUDIO;
> -
> - /* Wake up a process waiting for the queue being released.
> - * Note: There may be several processes waiting for a call
> - * to open() returning. */
> -
> - return rc;
> -}
> -
> -
> -static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd,
> - u_long arg)
> -{
> - u_long fmt;
> - int data;
> -#if 0
> - int size, nbufs;
> -#else
> - int size;
> -#endif
> -
> - switch (cmd) {
> - case SNDCTL_DSP_RESET:
> - sq_reset();
> - return 0;
> - case SNDCTL_DSP_POST:
> - case SNDCTL_DSP_SYNC:
> - return sq_fsync(file, file->f_path.dentry);
> -
> - /* ++TeSche: before changing any of these it's
> - * probably wise to wait until sound playing has
> - * settled down. */
> - case SNDCTL_DSP_SPEED:
> - sq_fsync(file, file->f_path.dentry);
> - IOCTL_IN(arg, data);
> - return IOCTL_OUT(arg, sound_set_speed(data));
> - case SNDCTL_DSP_STEREO:
> - sq_fsync(file, file->f_path.dentry);
> - IOCTL_IN(arg, data);
> - return IOCTL_OUT(arg, sound_set_stereo(data));
> - case SOUND_PCM_WRITE_CHANNELS:
> - sq_fsync(file, file->f_path.dentry);
> - IOCTL_IN(arg, data);
> - return IOCTL_OUT(arg, sound_set_stereo(data-1)+1);
> - case SNDCTL_DSP_SETFMT:
> - sq_fsync(file, file->f_path.dentry);
> - IOCTL_IN(arg, data);
> - return IOCTL_OUT(arg, sound_set_format(data));
> - case SNDCTL_DSP_GETFMTS:
> - fmt = 0;
> - if (sound.trans_write) {
> - if (sound.trans_write->ct_ulaw)
> - fmt |= AFMT_MU_LAW;
> - if (sound.trans_write->ct_alaw)
> - fmt |= AFMT_A_LAW;
> - if (sound.trans_write->ct_s8)
> - fmt |= AFMT_S8;
> - if (sound.trans_write->ct_u8)
> - fmt |= AFMT_U8;
> - if (sound.trans_write->ct_s16be)
> - fmt |= AFMT_S16_BE;
> - if (sound.trans_write->ct_u16be)
> - fmt |= AFMT_U16_BE;
> - if (sound.trans_write->ct_s16le)
> - fmt |= AFMT_S16_LE;
> - if (sound.trans_write->ct_u16le)
> - fmt |= AFMT_U16_LE;
> - }
> - return IOCTL_OUT(arg, fmt);
> - case SNDCTL_DSP_GETBLKSIZE:
> - size = sq.block_size
> - * sound.soft.size * (sound.soft.stereo + 1)
> - / (sound.hard.size * (sound.hard.stereo + 1));
> - return IOCTL_OUT(arg, size);
> - case SNDCTL_DSP_SUBDIVIDE:
> - break;
> -#if 0 /* Sorry can't do this at the moment. The CPM allocated buffers
> - * long ago that can't be changed.
> - */
> - case SNDCTL_DSP_SETFRAGMENT:
> - if (sq.count || sq.active || sq.syncing)
> - return -EINVAL;
> - IOCTL_IN(arg, size);
> - nbufs = size >> 16;
> - if (nbufs < 2 || nbufs > numBufs)
> - nbufs = numBufs;
> - size &= 0xffff;
> - if (size >= 8 && size <= 30) {
> - size = 1 << size;
> - size *= sound.hard.size * (sound.hard.stereo + 1);
> - size /= sound.soft.size * (sound.soft.stereo + 1);
> - if (size > (bufSize << 10))
> - size = bufSize << 10;
> - } else
> - size = bufSize << 10;
> - sq_setup(numBufs, size, sound_buffers);
> - sq.max_active = nbufs;
> - return 0;
> -#endif
> -
> - default:
> - return mixer_ioctl(inode, file, cmd, arg);
> - }
> - return -EINVAL;
> -}
> -
> -
> -
> -static const struct file_operations sq_fops =
> -{
> - .owner = THIS_MODULE,
> - .llseek = sound_lseek,
> - .read = sq_read, /* sq_read */
> - .write = sq_write,
> - .ioctl = sq_ioctl,
> - .open = sq_open,
> - .release = sq_release,
> -};
> -
> -
> -static void __init sq_init(void)
> -{
> - sq_unit = register_sound_dsp(&sq_fops, -1);
> - if (sq_unit < 0)
> - return;
> -
> - init_waitqueue_head(&sq.action_queue);
> - init_waitqueue_head(&sq.open_queue);
> - init_waitqueue_head(&sq.sync_queue);
> - init_waitqueue_head(&read_sq.action_queue);
> - init_waitqueue_head(&read_sq.open_queue);
> - init_waitqueue_head(&read_sq.sync_queue);
> -
> - sq.busy = 0;
> - read_sq.busy = 0;
> -
> - /* whatever you like as startup mode for /dev/dsp,
> - * (/dev/audio hasn't got a startup mode). note that
> - * once changed a new open() will *not* restore these!
> - */
> - sound.dsp.format = AFMT_S16_BE;
> - sound.dsp.stereo = 1;
> - sound.dsp.size = 16;
> -
> - /* set minimum rate possible without expanding */
> - sound.dsp.speed = 8000;
> -
> - /* before the first open to /dev/dsp this wouldn't be set */
> - sound.soft = sound.dsp;
> - sound.hard = sound.dsp;
> -
> - sound_silence();
> -}
> -
> -/*
> - * /dev/sndstat
> - */
> -
> -
> -/* state.buf should not overflow! */
> -
> -static int state_open(struct inode *inode, struct file *file)
> -{
> - char *buffer = state.buf, *mach = "", cs4218_buf[50];
> - int len = 0;
> -
> - if (state.busy)
> - return -EBUSY;
> -
> - state.ptr = 0;
> - state.busy = 1;
> -
> - sprintf(cs4218_buf, "Crystal CS4218 on TDM, ");
> - mach = cs4218_buf;
> -
> - len += sprintf(buffer+len, "%sDMA sound driver:\n", mach);
> -
> - len += sprintf(buffer+len, "\tsound.format = 0x%x", sound.soft.format);
> - switch (sound.soft.format) {
> - case AFMT_MU_LAW:
> - len += sprintf(buffer+len, " (mu-law)");
> - break;
> - case AFMT_A_LAW:
> - len += sprintf(buffer+len, " (A-law)");
> - break;
> - case AFMT_U8:
> - len += sprintf(buffer+len, " (unsigned 8 bit)");
> - break;
> - case AFMT_S8:
> - len += sprintf(buffer+len, " (signed 8 bit)");
> - break;
> - case AFMT_S16_BE:
> - len += sprintf(buffer+len, " (signed 16 bit big)");
> - break;
> - case AFMT_U16_BE:
> - len += sprintf(buffer+len, " (unsigned 16 bit big)");
> - break;
> - case AFMT_S16_LE:
> - len += sprintf(buffer+len, " (signed 16 bit little)");
> - break;
> - case AFMT_U16_LE:
> - len += sprintf(buffer+len, " (unsigned 16 bit little)");
> - break;
> - }
> - len += sprintf(buffer+len, "\n");
> - len += sprintf(buffer+len, "\tsound.speed = %dHz (phys. %dHz)\n",
> - sound.soft.speed, sound.hard.speed);
> - len += sprintf(buffer+len, "\tsound.stereo = 0x%x (%s)\n",
> - sound.soft.stereo, sound.soft.stereo ? "stereo" : "mono");
> - len += sprintf(buffer+len, "\tsq.block_size = %d sq.max_count = %d"
> - " sq.max_active = %d\n",
> - sq.block_size, sq.max_count, sq.max_active);
> - len += sprintf(buffer+len, "\tsq.count = %d sq.rear_size = %d\n", sq.count,
> - sq.rear_size);
> - len += sprintf(buffer+len, "\tsq.active = %d sq.syncing = %d\n",
> - sq.active, sq.syncing);
> - state.len = len;
> - return nonseekable_open(inode, file);
> -}
> -
> -
> -static int state_release(struct inode *inode, struct file *file)
> -{
> - state.busy = 0;
> - return 0;
> -}
> -
> -
> -static ssize_t state_read(struct file *file, char *buf, size_t count,
> - loff_t *ppos)
> -{
> - int n = state.len - state.ptr;
> - if (n > count)
> - n = count;
> - if (n <= 0)
> - return 0;
> - if (copy_to_user(buf, &state.buf[state.ptr], n))
> - return -EFAULT;
> - state.ptr += n;
> - return n;
> -}
> -
> -
> -static const struct file_operations state_fops =
> -{
> - .owner = THIS_MODULE,
> - .llseek = sound_lseek,
> - .read = state_read,
> - .open = state_open,
> - .release = state_release,
> -};
> -
> -
> -static void __init state_init(void)
> -{
> - state_unit = register_sound_special(&state_fops, SND_DEV_STATUS);
> - if (state_unit < 0)
> - return;
> - state.busy = 0;
> -}
> -
> -
> -/*** Common stuff ********************************************************/
> -
> -static long long sound_lseek(struct file *file, long long offset, int orig)
> -{
> - return -ESPIPE;
> -}
> -
> -
> -/*** Config & Setup **********************************************************/
> -
> -
> -int __init tdm8xx_sound_init(void)
> -{
> - int i, has_sound;
> - uint dp_offset;
> - volatile uint *sirp;
> - volatile cbd_t *bdp;
> - volatile cpm8xx_t *cp;
> - volatile smc_t *sp;
> - volatile smc_uart_t *up;
> - volatile immap_t *immap;
> -
> - has_sound = 0;
> -
> - /* Program the SI/TSA to use TDMa, connected to SMC2, for 4 bytes.
> - */
> - cp = cpmp; /* Get pointer to Communication Processor */
> - immap = (immap_t *)IMAP_ADDR; /* and to internal registers */
> -
> - /* Set all TDMa control bits to zero. This enables most features
> - * we want.
> - */
> - cp->cp_simode &= ~0x00000fff;
> -
> - /* Enable common receive/transmit clock pins, use IDL format.
> - * Sync on falling edge, transmit rising clock, receive falling
> - * clock, delay 1 bit on both Tx and Rx. Common Tx/Rx clocks and
> - * sync.
> - * Connect SMC2 to TSA.
> - */
> - cp->cp_simode |= 0x80000141;
> -
> - /* Configure port A pins for TDMa operation.
> - * The RPX-Lite (MPC850/823) loses SMC2 when TDM is used.
> - */
> - immap->im_ioport.iop_papar |= 0x01c0; /* Enable TDMa functions */
> - immap->im_ioport.iop_padir |= 0x00c0; /* Enable TDMa Tx/Rx */
> - immap->im_ioport.iop_padir &= ~0x0100; /* Enable L1RCLKa */
> -
> - immap->im_ioport.iop_pcpar |= 0x0800; /* Enable L1RSYNCa */
> - immap->im_ioport.iop_pcdir &= ~0x0800;
> -
> - /* Initialize the SI TDM routing table. We use TDMa only.
> - * The receive table and transmit table each have only one
> - * entry, to capture/send four bytes after each frame pulse.
> - * The 16-bit ram entry is 0000 0001 1000 1111. (SMC2)
> - */
> - cp->cp_sigmr = 0;
> - sirp = (uint *)cp->cp_siram;
> -
> - *sirp = 0x018f0000; /* Receive entry */
> - sirp += 64;
> - *sirp = 0x018f0000; /* Tramsmit entry */
> -
> - /* Enable single TDMa routing.
> - */
> - cp->cp_sigmr = 0x04;
> -
> - /* Initialize the SMC for transparent operation.
> - */
> - sp = &cpmp->cp_smc[1];
> - up = (smc_uart_t *)&cp->cp_dparam[PROFF_SMC2];
> -
> - /* We need to allocate a transmit and receive buffer
> - * descriptors from dual port ram.
> - */
> - dp_addr = cpm_dpalloc(sizeof(cbd_t) * numReadBufs, 8);
> -
> - /* Set the physical address of the host memory
> - * buffers in the buffer descriptors, and the
> - * virtual address for us to work with.
> - */
> - bdp = (cbd_t *)&cp->cp_dpmem[dp_addr];
> - up->smc_rbase = dp_offset;
> - rx_cur = rx_base = (cbd_t *)bdp;
> -
> - for (i=0; i<(numReadBufs-1); i++) {
> - bdp->cbd_bufaddr = 0;
> - bdp->cbd_datlen = 0;
> - bdp->cbd_sc = BD_SC_EMPTY | BD_SC_INTRPT;
> - bdp++;
> - }
> - bdp->cbd_bufaddr = 0;
> - bdp->cbd_datlen = 0;
> - bdp->cbd_sc = BD_SC_WRAP | BD_SC_EMPTY | BD_SC_INTRPT;
> -
> - /* Now, do the same for the transmit buffers.
> - */
> - dp_offset = cpm_dpalloc(sizeof(cbd_t) * numBufs, 8);
> -
> - bdp = (cbd_t *)&cp->cp_dpmem[dp_addr];
> - up->smc_tbase = dp_offset;
> - tx_cur = tx_base = (cbd_t *)bdp;
> -
> - for (i=0; i<(numBufs-1); i++) {
> - bdp->cbd_bufaddr = 0;
> - bdp->cbd_datlen = 0;
> - bdp->cbd_sc = BD_SC_INTRPT;
> - bdp++;
> - }
> - bdp->cbd_bufaddr = 0;
> - bdp->cbd_datlen = 0;
> - bdp->cbd_sc = (BD_SC_WRAP | BD_SC_INTRPT);
> -
> - /* Set transparent SMC mode.
> - * A few things are specific to our application. The codec interface
> - * is MSB first, hence the REVD selection. The CD/CTS pulse are
> - * used by the TSA to indicate the frame start to the SMC.
> - */
> - up->smc_rfcr = SCC_EB;
> - up->smc_tfcr = SCC_EB;
> - up->smc_mrblr = readbufSize * 1024;
> -
> - /* Set 16-bit reversed data, transparent mode.
> - */
> - sp->smc_smcmr = smcr_mk_clen(15) |
> - SMCMR_SM_TRANS | SMCMR_REVD | SMCMR_BS;
> -
> - /* Enable and clear events.
> - * Because of FIFO delays, all we need is the receive interrupt
> - * and we can process both the current receive and current
> - * transmit interrupt within a few microseconds of the transmit.
> - */
> - sp->smc_smce = 0xff;
> - sp->smc_smcm = SMCM_TXE | SMCM_TX | SMCM_RX;
> -
> - /* Send the CPM an initialize command.
> - */
> - cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
> - CPM_CR_INIT_TRX) | CPM_CR_FLG;
> - while (cp->cp_cpcr & CPM_CR_FLG);
> -
> - sound.mach = mach_cs4218;
> - has_sound = 1;
> -
> - /* Initialize beep stuff */
> - orig_mksound = kd_mksound;
> - kd_mksound = cs_mksound;
> - beep_buf = kmalloc(BEEP_BUFLEN * 4, GFP_KERNEL);
> - if (beep_buf == NULL)
> - printk(KERN_WARNING "dmasound: no memory for "
> - "beep buffer\n");
> -
> - if (!has_sound)
> - return -ENODEV;
> -
> - /* Initialize the software SPI.
> - */
> - sw_spi_init();
> -
> - /* Set up sound queue, /dev/audio and /dev/dsp. */
> -
> - /* Set default settings. */
> - sq_init();
> -
> - /* Set up /dev/sndstat. */
> - state_init();
> -
> - /* Set up /dev/mixer. */
> - mixer_init();
> -
> - if (!sound.mach.irqinit()) {
> - printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n");
> - return -ENODEV;
> - }
> -#ifdef MODULE
> - irq_installed = 1;
> -#endif
> -
> - printk(KERN_INFO "DMA sound driver installed, using %d buffers of %dk.\n",
> - numBufs, bufSize);
> -
> - return 0;
> -}
> -
> -/* Due to FIFOs and bit delays, the transmit interrupt occurs a few
> - * microseconds ahead of the receive interrupt.
> - * When we get an interrupt, we service the transmit first, then
> - * check for a receive to prevent the overhead of returning through
> - * the interrupt handler only to get back here right away during
> - * full duplex operation.
> - */
> -static void
> -cs4218_intr(void *dev_id)
> -{
> - volatile smc_t *sp;
> - volatile cpm8xx_t *cp;
> -
> - sp = &cpmp->cp_smc[1];
> -
> - if (sp->smc_smce & SCCM_TX) {
> - sp->smc_smce = SCCM_TX;
> - cs4218_tdm_tx_intr((void *)sp);
> - }
> -
> - if (sp->smc_smce & SCCM_RX) {
> - sp->smc_smce = SCCM_RX;
> - cs4218_tdm_rx_intr((void *)sp);
> - }
> -
> - if (sp->smc_smce & SCCM_TXE) {
> - /* Transmit underrun. This happens with the application
> - * didn't keep up sending buffers. We tell the SMC to
> - * restart, which will cause it to poll the current (next)
> - * BD. If the user supplied data since this occurred,
> - * we just start running again. If they didn't, the SMC
> - * will poll the descriptor until data is placed there.
> - */
> - sp->smc_smce = SCCM_TXE;
> - cp = cpmp; /* Get pointer to Communication Processor */
> - cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
> - CPM_CR_RESTART_TX) | CPM_CR_FLG;
> - while (cp->cp_cpcr & CPM_CR_FLG);
> - }
> -}
> -
> -
> -#define MAXARGS 8 /* Should be sufficient for now */
> -
> -void __init dmasound_setup(char *str, int *ints)
> -{
> - /* check the bootstrap parameter for "dmasound=" */
> -
> - switch (ints[0]) {
> - case 3:
> - if ((ints[3] < 0) || (ints[3] > MAX_CATCH_RADIUS))
> - printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius);
> - else
> - catchRadius = ints[3];
> - /* fall through */
> - case 2:
> - if (ints[1] < MIN_BUFFERS)
> - printk("dmasound_setup: invalid number of buffers, using default = %d\n", numBufs);
> - else
> - numBufs = ints[1];
> - if (ints[2] < MIN_BUFSIZE || ints[2] > MAX_BUFSIZE)
> - printk("dmasound_setup: invalid buffer size, using default = %d\n", bufSize);
> - else
> - bufSize = ints[2];
> - break;
> - case 0:
> - break;
> - default:
> - printk("dmasound_setup: invalid number of arguments\n");
> - }
> -}
> -
> -/* Software SPI functions.
> - * These are on Port B.
> - */
> -#define PB_SPICLK ((uint)0x00000002)
> -#define PB_SPIMOSI ((uint)0x00000004)
> -#define PB_SPIMISO ((uint)0x00000008)
> -
> -static
> -void sw_spi_init(void)
> -{
> - volatile cpm8xx_t *cp;
> - volatile uint *hcsr4;
> -
> - hcsr4 = (volatile uint *)HIOX_CSR4_ADDR;
> - cp = cpmp; /* Get pointer to Communication Processor */
> -
> - *hcsr4 &= ~HIOX_CSR4_AUDSPISEL; /* Disable SPI select */
> -
> - /* Make these Port B signals general purpose I/O.
> - * First, make sure the clock is low.
> - */
> - cp->cp_pbdat &= ~PB_SPICLK;
> - cp->cp_pbpar &= ~(PB_SPICLK | PB_SPIMOSI | PB_SPIMISO);
> -
> - /* Clock and Master Output are outputs.
> - */
> - cp->cp_pbdir |= (PB_SPICLK | PB_SPIMOSI);
> -
> - /* Master Input.
> - */
> - cp->cp_pbdir &= ~PB_SPIMISO;
> -
> -}
> -
> -/* Write the CS4218 control word out the SPI port. While the
> - * the control word is going out, the status word is arriving.
> - */
> -static
> -uint cs4218_ctl_write(uint ctlreg)
> -{
> - uint status;
> -
> - sw_spi_io((u_char *)&ctlreg, (u_char *)&status, 4);
> -
> - /* Shadow the control register.....I guess we could do
> - * the same for the status, but for now we just return it
> - * and let the caller decide.
> - */
> - cs4218_control = ctlreg;
> - return status;
> -}
> -
> -static
> -void sw_spi_io(u_char *obuf, u_char *ibuf, uint bcnt)
> -{
> - int bits, i;
> - u_char outbyte, inbyte;
> - volatile cpm8xx_t *cp;
> - volatile uint *hcsr4;
> -
> - hcsr4 = (volatile uint *)HIOX_CSR4_ADDR;
> - cp = cpmp; /* Get pointer to Communication Processor */
> -
> - /* The timing on the bus is pretty slow. Code inefficiency
> - * and eieio() is our friend here :-).
> - */
> - cp->cp_pbdat &= ~PB_SPICLK;
> - *hcsr4 |= HIOX_CSR4_AUDSPISEL; /* Enable SPI select */
> - eieio();
> -
> - /* Clock in/out the bytes. Data is valid on the falling edge
> - * of the clock. Data is MSB first.
> - */
> - for (i=0; i<bcnt; i++) {
> - outbyte = *obuf++;
> - inbyte = 0;
> - for (bits=0; bits<8; bits++) {
> - eieio();
> - cp->cp_pbdat |= PB_SPICLK;
> - eieio();
> - if (outbyte & 0x80)
> - cp->cp_pbdat |= PB_SPIMOSI;
> - else
> - cp->cp_pbdat &= ~PB_SPIMOSI;
> - eieio();
> - cp->cp_pbdat &= ~PB_SPICLK;
> - eieio();
> - outbyte <<= 1;
> - inbyte <<= 1;
> - if (cp->cp_pbdat & PB_SPIMISO)
> - inbyte |= 1;
> - }
> - *ibuf++ = inbyte;
> - }
> -
> - *hcsr4 &= ~HIOX_CSR4_AUDSPISEL; /* Disable SPI select */
> - eieio();
> -}
> -
> -void cleanup_module(void)
> -{
> - if (irq_installed) {
> - sound_silence();
> -#ifdef MODULE
> - sound.mach.irqcleanup();
> -#endif
> - }
> -
> - sq_release_read_buffers();
> - sq_release_buffers();
> -
> - if (mixer_unit >= 0)
> - unregister_sound_mixer(mixer_unit);
> - if (state_unit >= 0)
> - unregister_sound_special(state_unit);
> - if (sq_unit >= 0)
> - unregister_sound_dsp(sq_unit);
> -}
> -
> -module_init(tdm8xx_sound_init);
> -module_exit(cleanup_module);
> -
> -
-
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