diff options
Diffstat (limited to 'sound/oss/dmasound')
| -rw-r--r-- | sound/oss/dmasound/Kconfig | 58 | ||||
| -rw-r--r-- | sound/oss/dmasound/Makefile | 13 | ||||
| -rw-r--r-- | sound/oss/dmasound/awacs_defs.h | 251 | ||||
| -rw-r--r-- | sound/oss/dmasound/dac3550a.c | 210 | ||||
| -rw-r--r-- | sound/oss/dmasound/dmasound.h | 277 | ||||
| -rw-r--r-- | sound/oss/dmasound/dmasound_atari.c | 1600 | ||||
| -rw-r--r-- | sound/oss/dmasound/dmasound_awacs.c | 3176 | ||||
| -rw-r--r-- | sound/oss/dmasound/dmasound_core.c | 1829 | ||||
| -rw-r--r-- | sound/oss/dmasound/dmasound_paula.c | 743 | ||||
| -rw-r--r-- | sound/oss/dmasound/dmasound_q40.c | 634 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas3001c.c | 850 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas3001c.h | 64 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas3001c_tables.c | 375 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas3004.c | 1140 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas3004.h | 77 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas3004_tables.c | 301 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas_common.c | 214 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas_common.h | 284 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas_eq_prefs.h | 24 | ||||
| -rw-r--r-- | sound/oss/dmasound/tas_ioctl.h | 24 | ||||
| -rw-r--r-- | sound/oss/dmasound/trans_16.c | 897 |
21 files changed, 13041 insertions, 0 deletions
diff --git a/sound/oss/dmasound/Kconfig b/sound/oss/dmasound/Kconfig new file mode 100644 index 0000000..cb84558 --- /dev/null +++ b/sound/oss/dmasound/Kconfig @@ -0,0 +1,58 @@ +config DMASOUND_ATARI + tristate "Atari DMA sound support" + depends on ATARI && SOUND + select DMASOUND + help + If you want to use the internal audio of your Atari in Linux, answer + Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND_PMAC + tristate "PowerMac DMA sound support" + depends on PPC32 && PPC_PMAC && SOUND && I2C + select DMASOUND + help + If you want to use the internal audio of your PowerMac in Linux, + answer Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND_PAULA + tristate "Amiga DMA sound support" + depends on (AMIGA || APUS) && SOUND + select DMASOUND + help + If you want to use the internal audio of your Amiga in Linux, answer + Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND_Q40 + tristate "Q40 sound support" + depends on Q40 && SOUND + select DMASOUND + help + If you want to use the internal audio of your Q40 in Linux, answer + Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND + tristate diff --git a/sound/oss/dmasound/Makefile b/sound/oss/dmasound/Makefile new file mode 100644 index 0000000..4611636 --- /dev/null +++ b/sound/oss/dmasound/Makefile @@ -0,0 +1,13 @@ +# +# Makefile for the DMA sound driver +# + +dmasound_pmac-y += dmasound_awacs.o \ + trans_16.o dac3550a.o tas_common.o \ + tas3001c.o tas3001c_tables.o \ + tas3004.o tas3004_tables.o + +obj-$(CONFIG_DMASOUND_ATARI) += dmasound_core.o dmasound_atari.o +obj-$(CONFIG_DMASOUND_PMAC) += dmasound_core.o dmasound_pmac.o +obj-$(CONFIG_DMASOUND_PAULA) += dmasound_core.o dmasound_paula.o +obj-$(CONFIG_DMASOUND_Q40) += dmasound_core.o dmasound_q40.o diff --git a/sound/oss/dmasound/awacs_defs.h b/sound/oss/dmasound/awacs_defs.h new file mode 100644 index 0000000..2194f46 --- /dev/null +++ b/sound/oss/dmasound/awacs_defs.h @@ -0,0 +1,251 @@ +/*********************************************************/ +/* This file was written by someone, somewhere, sometime */ +/* And is released into the Public Domain */ +/*********************************************************/ + +#ifndef _AWACS_DEFS_H_ +#define _AWACS_DEFS_H_ + +/*******************************/ +/* AWACs Audio Register Layout */ +/*******************************/ + +struct awacs_regs { + unsigned control; /* Audio control register */ + unsigned pad0[3]; + unsigned codec_ctrl; /* Codec control register */ + unsigned pad1[3]; + unsigned codec_stat; /* Codec status register */ + unsigned pad2[3]; + unsigned clip_count; /* Clipping count register */ + unsigned pad3[3]; + unsigned byteswap; /* Data is little-endian if 1 */ +}; + +/*******************/ +/* Audio Bit Masks */ +/*******************/ + +/* Audio Control Reg Bit Masks */ +/* ----- ------- --- --- ----- */ +#define MASK_ISFSEL (0xf) /* Input SubFrame Select */ +#define MASK_OSFSEL (0xf << 4) /* Output SubFrame Select */ +#define MASK_RATE (0x7 << 8) /* Sound Rate */ +#define MASK_CNTLERR (0x1 << 11) /* Error */ +#define MASK_PORTCHG (0x1 << 12) /* Port Change */ +#define MASK_IEE (0x1 << 13) /* Enable Interrupt on Error */ +#define MASK_IEPC (0x1 << 14) /* Enable Interrupt on Port Change */ +#define MASK_SSFSEL (0x3 << 15) /* Status SubFrame Select */ + +/* Audio Codec Control Reg Bit Masks */ +/* ----- ----- ------- --- --- ----- */ +#define MASK_NEWECMD (0x1 << 24) /* Lock: don't write to reg when 1 */ +#define MASK_EMODESEL (0x3 << 22) /* Send info out on which frame? */ +#define MASK_EXMODEADDR (0x3ff << 12) /* Extended Mode Address -- 10 bits */ +#define MASK_EXMODEDATA (0xfff) /* Extended Mode Data -- 12 bits */ + +/* Audio Codec Control Address Values / Masks */ +/* ----- ----- ------- ------- ------ - ----- */ +#define MASK_ADDR0 (0x0 << 12) /* Expanded Data Mode Address 0 */ +#define MASK_ADDR_MUX MASK_ADDR0 /* Mux Control */ +#define MASK_ADDR_GAIN MASK_ADDR0 + +#define MASK_ADDR1 (0x1 << 12) /* Expanded Data Mode Address 1 */ +#define MASK_ADDR_MUTE MASK_ADDR1 +#define MASK_ADDR_RATE MASK_ADDR1 + +#define MASK_ADDR2 (0x2 << 12) /* Expanded Data Mode Address 2 */ +#define MASK_ADDR_VOLA MASK_ADDR2 /* Volume Control A -- Headphones */ +#define MASK_ADDR_VOLHD MASK_ADDR2 + +#define MASK_ADDR4 (0x4 << 12) /* Expanded Data Mode Address 4 */ +#define MASK_ADDR_VOLC MASK_ADDR4 /* Volume Control C -- Speaker */ +#define MASK_ADDR_VOLSPK MASK_ADDR4 + +/* additional registers of screamer */ +#define MASK_ADDR5 (0x5 << 12) /* Expanded Data Mode Address 5 */ +#define MASK_ADDR6 (0x6 << 12) /* Expanded Data Mode Address 6 */ +#define MASK_ADDR7 (0x7 << 12) /* Expanded Data Mode Address 7 */ + +/* Address 0 Bit Masks & Macros */ +/* ------- - --- ----- - ------ */ +#define MASK_GAINRIGHT (0xf) /* Gain Right Mask */ +#define MASK_GAINLEFT (0xf << 4) /* Gain Left Mask */ +#define MASK_GAINLINE (0x1 << 8) /* Disable Mic preamp */ +#define MASK_GAINMIC (0x0 << 8) /* Enable Mic preamp */ + +#define MASK_MUX_CD (0x1 << 9) /* Select CD in MUX */ +#define MASK_MUX_MIC (0x1 << 10) /* Select Mic in MUX */ +#define MASK_MUX_AUDIN (0x1 << 11) /* Select Audio In in MUX */ +#define MASK_MUX_LINE MASK_MUX_AUDIN + +#define GAINRIGHT(x) ((x) & MASK_GAINRIGHT) +#define GAINLEFT(x) (((x) << 4) & MASK_GAINLEFT) + +#define DEF_CD_GAIN 0x00bb +#define DEF_MIC_GAIN 0x00cc + +/* Address 1 Bit Masks */ +/* ------- - --- ----- */ +#define MASK_ADDR1RES1 (0x3) /* Reserved */ +#define MASK_RECALIBRATE (0x1 << 2) /* Recalibrate */ +#define MASK_SAMPLERATE (0x7 << 3) /* Sample Rate: */ +#define MASK_LOOPTHRU (0x1 << 6) /* Loopthrough Enable */ +#define MASK_CMUTE (0x1 << 7) /* Output C (Speaker) Mute when 1 */ +#define MASK_SPKMUTE MASK_CMUTE +#define MASK_ADDR1RES2 (0x1 << 8) /* Reserved */ +#define MASK_AMUTE (0x1 << 9) /* Output A (Headphone) Mute when 1 */ +#define MASK_HDMUTE MASK_AMUTE +#define MASK_PAROUT0 (0x1 << 10) /* Parallel Output 0 */ +#define MASK_PAROUT1 (0x2 << 10) /* Parallel Output 1 */ + +#define MASK_MIC_BOOST (0x4) /* screamer mic boost */ + +#define SAMPLERATE_48000 (0x0 << 3) /* 48 or 44.1 kHz */ +#define SAMPLERATE_32000 (0x1 << 3) /* 32 or 29.4 kHz */ +#define SAMPLERATE_24000 (0x2 << 3) /* 24 or 22.05 kHz */ +#define SAMPLERATE_19200 (0x3 << 3) /* 19.2 or 17.64 kHz */ +#define SAMPLERATE_16000 (0x4 << 3) /* 16 or 14.7 kHz */ +#define SAMPLERATE_12000 (0x5 << 3) /* 12 or 11.025 kHz */ +#define SAMPLERATE_9600 (0x6 << 3) /* 9.6 or 8.82 kHz */ +#define SAMPLERATE_8000 (0x7 << 3) /* 8 or 7.35 kHz */ + +/* Address 2 & 4 Bit Masks & Macros */ +/* ------- - - - --- ----- - ------ */ +#define MASK_OUTVOLRIGHT (0xf) /* Output Right Volume */ +#define MASK_ADDR2RES1 (0x2 << 4) /* Reserved */ +#define MASK_ADDR4RES1 MASK_ADDR2RES1 +#define MASK_OUTVOLLEFT (0xf << 6) /* Output Left Volume */ +#define MASK_ADDR2RES2 (0x2 << 10) /* Reserved */ +#define MASK_ADDR4RES2 MASK_ADDR2RES2 + +#define VOLRIGHT(x) (((~(x)) & MASK_OUTVOLRIGHT)) +#define VOLLEFT(x) (((~(x)) << 6) & MASK_OUTVOLLEFT) + +/* Audio Codec Status Reg Bit Masks */ +/* ----- ----- ------ --- --- ----- */ +#define MASK_EXTEND (0x1 << 23) /* Extend */ +#define MASK_VALID (0x1 << 22) /* Valid Data? */ +#define MASK_OFLEFT (0x1 << 21) /* Overflow Left */ +#define MASK_OFRIGHT (0x1 << 20) /* Overflow Right */ +#define MASK_ERRCODE (0xf << 16) /* Error Code */ +#define MASK_REVISION (0xf << 12) /* Revision Number */ +#define MASK_MFGID (0xf << 8) /* Mfg. ID */ +#define MASK_CODSTATRES (0xf << 4) /* bits 4 - 7 reserved */ +#define MASK_INPPORT (0xf) /* Input Port */ +#define MASK_HDPCONN 8 /* headphone plugged in */ + +/* Clipping Count Reg Bit Masks */ +/* -------- ----- --- --- ----- */ +#define MASK_CLIPLEFT (0xff << 7) /* Clipping Count, Left Channel */ +#define MASK_CLIPRIGHT (0xff) /* Clipping Count, Right Channel */ + +/* DBDMA ChannelStatus Bit Masks */ +/* ----- ------------- --- ----- */ +#define MASK_CSERR (0x1 << 7) /* Error */ +#define MASK_EOI (0x1 << 6) /* End of Input -- only for Input Channel */ +#define MASK_CSUNUSED (0x1f << 1) /* bits 1-5 not used */ +#define MASK_WAIT (0x1) /* Wait */ + +/* Various Rates */ +/* ------- ----- */ +#define RATE_48000 (0x0 << 8) /* 48 kHz */ +#define RATE_44100 (0x0 << 8) /* 44.1 kHz */ +#define RATE_32000 (0x1 << 8) /* 32 kHz */ +#define RATE_29400 (0x1 << 8) /* 29.4 kHz */ +#define RATE_24000 (0x2 << 8) /* 24 kHz */ +#define RATE_22050 (0x2 << 8) /* 22.05 kHz */ +#define RATE_19200 (0x3 << 8) /* 19.2 kHz */ +#define RATE_17640 (0x3 << 8) /* 17.64 kHz */ +#define RATE_16000 (0x4 << 8) /* 16 kHz */ +#define RATE_14700 (0x4 << 8) /* 14.7 kHz */ +#define RATE_12000 (0x5 << 8) /* 12 kHz */ +#define RATE_11025 (0x5 << 8) /* 11.025 kHz */ +#define RATE_9600 (0x6 << 8) /* 9.6 kHz */ +#define RATE_8820 (0x6 << 8) /* 8.82 kHz */ +#define RATE_8000 (0x7 << 8) /* 8 kHz */ +#define RATE_7350 (0x7 << 8) /* 7.35 kHz */ + +#define RATE_LOW 1 /* HIGH = 48kHz, etc; LOW = 44.1kHz, etc. */ + +/*******************/ +/* Burgundy values */ +/*******************/ + +#define MASK_ADDR_BURGUNDY_INPSEL21 (0x11 << 12) +#define MASK_ADDR_BURGUNDY_INPSEL3 (0x12 << 12) + +#define MASK_ADDR_BURGUNDY_GAINCH1 (0x13 << 12) +#define MASK_ADDR_BURGUNDY_GAINCH2 (0x14 << 12) +#define MASK_ADDR_BURGUNDY_GAINCH3 (0x15 << 12) +#define MASK_ADDR_BURGUNDY_GAINCH4 (0x16 << 12) + +#define MASK_ADDR_BURGUNDY_VOLCH1 (0x20 << 12) +#define MASK_ADDR_BURGUNDY_VOLCH2 (0x21 << 12) +#define MASK_ADDR_BURGUNDY_VOLCH3 (0x22 << 12) +#define MASK_ADDR_BURGUNDY_VOLCH4 (0x23 << 12) + +#define MASK_ADDR_BURGUNDY_OUTPUTSELECTS (0x2B << 12) +#define MASK_ADDR_BURGUNDY_OUTPUTENABLES (0x2F << 12) + +#define MASK_ADDR_BURGUNDY_MASTER_VOLUME (0x30 << 12) + +#define MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES (0x60 << 12) + +#define MASK_ADDR_BURGUNDY_ATTENSPEAKER (0x62 << 12) +#define MASK_ADDR_BURGUNDY_ATTENLINEOUT (0x63 << 12) +#define MASK_ADDR_BURGUNDY_ATTENHP (0x64 << 12) + +#define MASK_ADDR_BURGUNDY_VOLCD (MASK_ADDR_BURGUNDY_VOLCH1) +#define MASK_ADDR_BURGUNDY_VOLLINE (MASK_ADDR_BURGUNDY_VOLCH2) +#define MASK_ADDR_BURGUNDY_VOLMIC (MASK_ADDR_BURGUNDY_VOLCH3) +#define MASK_ADDR_BURGUNDY_VOLMODEM (MASK_ADDR_BURGUNDY_VOLCH4) + +#define MASK_ADDR_BURGUNDY_GAINCD (MASK_ADDR_BURGUNDY_GAINCH1) +#define MASK_ADDR_BURGUNDY_GAINLINE (MASK_ADDR_BURGUNDY_GAINCH2) +#define MASK_ADDR_BURGUNDY_GAINMIC (MASK_ADDR_BURGUNDY_GAINCH3) +#define MASK_ADDR_BURGUNDY_GAINMODEM (MASK_ADDR_BURGUNDY_VOLCH4) + + +/* These are all default values for the burgundy */ +#define DEF_BURGUNDY_INPSEL21 (0xAA) +#define DEF_BURGUNDY_INPSEL3 (0x0A) + +#define DEF_BURGUNDY_GAINCD (0x33) +#define DEF_BURGUNDY_GAINLINE (0x44) +#define DEF_BURGUNDY_GAINMIC (0x44) +#define DEF_BURGUNDY_GAINMODEM (0x06) + +/* Remember: lowest volume here is 0x9b */ +#define DEF_BURGUNDY_VOLCD (0xCCCCCCCC) +#define DEF_BURGUNDY_VOLLINE (0x00000000) +#define DEF_BURGUNDY_VOLMIC (0x00000000) +#define DEF_BURGUNDY_VOLMODEM (0xCCCCCCCC) + +#define DEF_BURGUNDY_OUTPUTSELECTS (0x010f010f) +#define DEF_BURGUNDY_OUTPUTENABLES (0x0A) + +#define DEF_BURGUNDY_MASTER_VOLUME (0xFFFFFFFF) + +#define DEF_BURGUNDY_MORE_OUTPUTENABLES (0x7E) + +#define DEF_BURGUNDY_ATTENSPEAKER (0x44) +#define DEF_BURGUNDY_ATTENLINEOUT (0xCC) +#define DEF_BURGUNDY_ATTENHP (0xCC) + +/*********************/ +/* i2s layout values */ +/*********************/ + +#define I2S_REG_INT_CTL 0x00 +#define I2S_REG_SERIAL_FORMAT 0x10 +#define I2S_REG_CODEC_MSG_OUT 0x20 +#define I2S_REG_CODEC_MSG_IN 0x30 +#define I2S_REG_FRAME_COUNT 0x40 +#define I2S_REG_FRAME_MATCH 0x50 +#define I2S_REG_DATAWORD_SIZES 0x60 +#define I2S_REG_PEAKLEVEL_SEL 0x70 +#define I2S_REG_PEAKLEVEL_IN0 0x80 +#define I2S_REG_PEAKLEVEL_IN1 0x90 + +#endif /* _AWACS_DEFS_H_ */ diff --git a/sound/oss/dmasound/dac3550a.c b/sound/oss/dmasound/dac3550a.c new file mode 100644 index 0000000..533895e --- /dev/null +++ b/sound/oss/dmasound/dac3550a.c @@ -0,0 +1,210 @@ +/* + * Driver for the i2c/i2s based DAC3550a sound chip used + * on some Apple iBooks. Also known as "DACA". + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/proc_fs.h> +#include <linux/ioport.h> +#include <linux/sysctl.h> +#include <linux/types.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <asm/uaccess.h> +#include <asm/errno.h> +#include <asm/io.h> + +#include "dmasound.h" + +/* FYI: This code was derived from the tas3001c.c Texas/Tumbler mixer + * control code, as well as info derived from the AppleDACAAudio driver + * from Darwin CVS (main thing I derived being register numbers and + * values, as well as when to make the calls). */ + +#define I2C_DRIVERID_DACA (0xFDCB) + +#define DACA_VERSION "0.1" +#define DACA_DATE "20010930" + +static int cur_left_vol; +static int cur_right_vol; +static struct i2c_client *daca_client; + +static int daca_attach_adapter(struct i2c_adapter *adapter); +static int daca_detect_client(struct i2c_adapter *adapter, int address); +static int daca_detach_client(struct i2c_client *client); + +struct i2c_driver daca_driver = { + .owner = THIS_MODULE, + .name = "DAC3550A driver V " DACA_VERSION, + .id = I2C_DRIVERID_DACA, + .flags = I2C_DF_NOTIFY, + .attach_adapter = daca_attach_adapter, + .detach_client = daca_detach_client, +}; + +#define VOL_MAX ((1<<20) - 1) + +void daca_get_volume(uint * left_vol, uint *right_vol) +{ + *left_vol = cur_left_vol >> 5; + *right_vol = cur_right_vol >> 5; +} + +int daca_set_volume(uint left_vol, uint right_vol) +{ + unsigned short voldata; + + if (!daca_client) + return -1; + + /* Derived from experience, not from any specific values */ + left_vol <<= 5; + right_vol <<= 5; + + if (left_vol > VOL_MAX) + left_vol = VOL_MAX; + if (right_vol > VOL_MAX) + right_vol = VOL_MAX; + + voldata = ((left_vol >> 14) & 0x3f) << 8; + voldata |= (right_vol >> 14) & 0x3f; + + if (i2c_smbus_write_word_data(daca_client, 2, voldata) < 0) { + printk("daca: failed to set volume \n"); + return -1; + } + + cur_left_vol = left_vol; + cur_right_vol = right_vol; + + return 0; +} + +int daca_leave_sleep(void) +{ + if (!daca_client) + return -1; + + /* Do a short sleep, just to make sure I2C bus is awake and paying + * attention to us + */ + msleep(20); + /* Write the sample rate reg the value it needs */ + i2c_smbus_write_byte_data(daca_client, 1, 8); + daca_set_volume(cur_left_vol >> 5, cur_right_vol >> 5); + /* Another short delay, just to make sure the other I2C bus writes + * have taken... + */ + msleep(20); + /* Write the global config reg - invert right power amp, + * DAC on, use 5-volt mode */ + i2c_smbus_write_byte_data(daca_client, 3, 0x45); + + return 0; +} + +int daca_enter_sleep(void) +{ + if (!daca_client) + return -1; + + i2c_smbus_write_byte_data(daca_client, 1, 8); + daca_set_volume(cur_left_vol >> 5, cur_right_vol >> 5); + + /* Write the global config reg - invert right power amp, + * DAC on, enter low-power mode, use 5-volt mode + */ + i2c_smbus_write_byte_data(daca_client, 3, 0x65); + + return 0; +} + +static int daca_attach_adapter(struct i2c_adapter *adapter) +{ + if (!strncmp(adapter->name, "mac-io", 6)) + daca_detect_client(adapter, 0x4d); + return 0; +} + +static int daca_init_client(struct i2c_client * new_client) +{ + /* + * Probe is not working with the current i2c-keywest + * driver. We try to use addr 0x4d on each adapters + * instead, by setting the format register. + * + * FIXME: I'm sure that can be obtained from the + * device-tree. --BenH. + */ + + /* Write the global config reg - invert right power amp, + * DAC on, use 5-volt mode + */ + if (i2c_smbus_write_byte_data(new_client, 3, 0x45)) + return -1; + + i2c_smbus_write_byte_data(new_client, 1, 8); + daca_client = new_client; + daca_set_volume(15000, 15000); + + return 0; +} + +static int daca_detect_client(struct i2c_adapter *adapter, int address) +{ + const char *client_name = "DAC 3550A Digital Equalizer"; + struct i2c_client *new_client; + int rc = -ENODEV; + + new_client = kmalloc(sizeof(*new_client), GFP_KERNEL); + if (!new_client) + return -ENOMEM; + memset(new_client, 0, sizeof(*new_client)); + + new_client->addr = address; + new_client->adapter = adapter; + new_client->driver = &daca_driver; + new_client->flags = 0; + strcpy(new_client->name, client_name); + + if (daca_init_client(new_client)) + goto bail; + + /* Tell the i2c layer a new client has arrived */ + if (i2c_attach_client(new_client)) + goto bail; + + return 0; + bail: + kfree(new_client); + return rc; +} + + +static int daca_detach_client(struct i2c_client *client) +{ + if (client == daca_client) + daca_client = NULL; + + i2c_detach_client(client); + kfree(client); + return 0; +} + +void daca_cleanup(void) +{ + i2c_del_driver(&daca_driver); +} + +int daca_init(void) +{ + printk("dac3550a driver version %s (%s)\n",DACA_VERSION,DACA_DATE); + return i2c_add_driver(&daca_driver); +} diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h new file mode 100644 index 0000000..9a2f50f --- /dev/null +++ b/sound/oss/dmasound/dmasound.h @@ -0,0 +1,277 @@ +#ifndef _dmasound_h_ +/* + * linux/sound/oss/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 _dmasound_h_ + +#include <linux/types.h> +#include <linux/config.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 __user *)(arg)); \ + if (error) return error; \ + } while (0) +#define IOCTL_OUT(arg, ret) ioctl_return((int __user *)(arg), ret) + +static inline int ioctl_return(int __user *addr, int value) +{ + return value < 0 ? value : put_user(value, addr); +} + + + /* + * Configuration + */ + +#undef HAS_8BIT_TABLES +#undef HAS_RECORD + +#if defined(CONFIG_DMASOUND_ATARI) || defined(CONFIG_DMASOUND_ATARI_MODULE) ||\ + defined(CONFIG_DMASOUND_PAULA) || defined(CONFIG_DMASOUND_PAULA_MODULE) ||\ + defined(CONFIG_DMASOUND_Q40) || defined(CONFIG_DMASOUND_Q40_MODULE) +#define HAS_8BIT_TABLES +#define MIN_BUFFERS 4 +#define MIN_BUFSIZE (1<<12) /* in bytes (- where does this come from ?) */ +#define MIN_FRAG_SIZE 8 /* not 100% sure about this */ +#define MAX_BUFSIZE (1<<17) /* Limit for Amiga is 128 kb */ +#define MAX_FRAG_SIZE 15 /* allow *4 for mono-8 => stereo-16 (for multi) */ + +#else /* is pmac and multi is off */ + +#define MIN_BUFFERS 2 +#define MIN_BUFSIZE (1<<8) /* in bytes */ +#define MIN_FRAG_SIZE 8 +#define MAX_BUFSIZE (1<<18) /* this is somewhat arbitrary for pmac */ +#define MAX_FRAG_SIZE 16 /* need to allow *4 for mono-8 => stereo-16 */ +#endif + +#define DEFAULT_N_BUFFERS 4 +#define DEFAULT_BUFF_SIZE (1<<15) + +#if defined(CONFIG_DMASOUND_PMAC) || defined(CONFIG_DMASOUND_PMAC_MODULE) +#define HAS_RECORD +#endif + + /* + * Initialization + */ + +extern int dmasound_init(void); +#ifdef MODULE +extern void dmasound_deinit(void); +#else +#define dmasound_deinit() do { } while (0) +#endif + +/* 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; + struct module *owner; + void *(*dma_alloc)(unsigned int, int); + 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 __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_alaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_s8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_u8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_s16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_u16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_s16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_u16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); +} TRANS; + +struct 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 */ +#ifdef HAS_RECORD + TRANS *trans_read; /* supported translations */ +#endif + 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 */ + spinlock_t lock; +}; + +extern struct sound_settings dmasound; + +#ifdef HAS_8BIT_TABLES +extern char dmasound_ulaw2dma8[]; +extern char dmasound_alaw2dma8[]; +#endif + + /* + * Mid level stuff + */ + +static inline int dmasound_set_volume(int volume) +{ + return dmasound.mach.setVolume(volume); +} + +static inline int dmasound_set_bass(int bass) +{ + return dmasound.mach.setBass ? dmasound.mach.setBass(bass) : 50; +} + +static inline int dmasound_set_treble(int treble) +{ + return dmasound.mach.setTreble ? dmasound.mach.setTreble(treble) : 50; +} + +static inline int dmasound_set_gain(int gain) +{ + return dmasound.mach.setGain ? dmasound.mach.setGain(gain) : 100; +} + + + /* + * 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)) + +extern struct sound_queue dmasound_write_sq; +#define write_sq dmasound_write_sq + +#ifdef HAS_RECORD +extern struct sound_queue dmasound_read_sq; +#define read_sq dmasound_read_sq +#endif + +extern int dmasound_catchRadius; +#define catchRadius dmasound_catchRadius + +/* define the value to be put in the byte-swap reg in mac-io + when we want it to swap for us. +*/ +#define BS_VAL 1 + +#define SW_INPUT_VOLUME_SCALE 4 +#define SW_INPUT_VOLUME_DEFAULT (128 / SW_INPUT_VOLUME_SCALE) + +extern int expand_bal; /* Balance factor for expanding (not volume!) */ +extern int expand_read_bal; /* Balance factor for reading */ +extern uint software_input_volume; /* software implemented recording volume! */ + +#endif /* _dmasound_h_ */ diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c new file mode 100644 index 0000000..8daaf87 --- /dev/null +++ b/sound/oss/dmasound/dmasound_atari.c @@ -0,0 +1,1600 @@ +/* + * linux/sound/oss/dmasound/dmasound_atari.c + * + * Atari TT and Falcon DMA Sound Driver + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits + * prior to 28/01/2001 + * + * 28/01/2001 [0.1] Iain Sandoe + * - added versioning + * - put in and populated the hardware_afmts field. + * [0.2] - put in SNDCTL_DSP_GETCAPS value. + * 01/02/2001 [0.3] - put in default hard/soft settings. + */ + + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <linux/mm.h> +#include <linux/spinlock.h> +#include <linux/interrupt.h> + +#include <asm/uaccess.h> +#include <asm/atariints.h> +#include <asm/atari_stram.h> + +#include "dmasound.h" + +#define DMASOUND_ATARI_REVISION 0 +#define DMASOUND_ATARI_EDITION 3 + +extern void atari_microwire_cmd(int cmd); + +static int is_falcon; +static int write_sq_ignore_int; /* ++TeSche: used for Falcon */ + +static int expand_bal; /* Balance factor for expanding (not volume!) */ +static int expand_data; /* Data for expanding */ + + +/*** 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 ata_ct_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_s16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_u16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_s16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_u16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_s16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_u16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_s16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_u16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); + + +/*** Low level stuff *********************************************************/ + + +static void *AtaAlloc(unsigned int size, int flags); +static void AtaFree(void *, unsigned int size); +static int AtaIrqInit(void); +#ifdef MODULE +static void AtaIrqCleanUp(void); +#endif /* MODULE */ +static int AtaSetBass(int bass); +static int AtaSetTreble(int treble); +static void TTSilence(void); +static void TTInit(void); +static int TTSetFormat(int format); +static int TTSetVolume(int volume); +static int TTSetGain(int gain); +static void FalconSilence(void); +static void FalconInit(void); +static int FalconSetFormat(int format); +static int FalconSetVolume(int volume); +static void AtaPlayNextFrame(int index); +static void AtaPlay(void); +static irqreturn_t AtaInterrupt(int irq, void *dummy, struct pt_regs *fp); + +/*** Mid level stuff *********************************************************/ + +static void TTMixerInit(void); +static void FalconMixerInit(void); +static int AtaMixerIoctl(u_int cmd, u_long arg); +static int TTMixerIoctl(u_int cmd, u_long arg); +static int FalconMixerIoctl(u_int cmd, u_long arg); +static int AtaWriteSqSetup(void); +static int AtaSqOpen(mode_t mode); +static int TTStateInfo(char *buffer, size_t space); +static int FalconStateInfo(char *buffer, size_t space); + + +/*** Translations ************************************************************/ + + +static ssize_t ata_ct_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8 + : dmasound_alaw2dma8; + ssize_t count, used; + u_char *p = &frame[*frameUsed]; + + count = min_t(unsigned long, userCount, frameLeft); + if (dmasound.soft.stereo) + count &= ~1; + used = count; + while (count > 0) { + u_char data; + if (get_user(data, userPtr++)) + return -EFAULT; + *p++ = table[data]; + count--; + } + *frameUsed += used; + return used; +} + + +static ssize_t ata_ct_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + void *p = &frame[*frameUsed]; + + count = min_t(unsigned long, userCount, frameLeft); + if (dmasound.soft.stereo) + count &= ~1; + used = count; + if (copy_from_user(p, userPtr, count)) + return -EFAULT; + *frameUsed += used; + return used; +} + + +static ssize_t ata_ct_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft); + used = count; + while (count > 0) { + u_char data; + if (get_user(data, userPtr++)) + return -EFAULT; + *p++ = data ^ 0x80; + count--; + } + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + *p++ = data ^ 0x8080; + count--; + } + } + *frameUsed += used; + return used; +} + + +static ssize_t ata_ct_s16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + *p++ = data; + *p++ = data; + count--; + } + *frameUsed += used*2; + } else { + void *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft) & ~3; + used = count; + if (copy_from_user(p, userPtr, count)) + return -EFAULT; + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ct_u16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data ^= 0x8000; + *p++ = data; + *p++ = data; + count--; + } + *frameUsed += used*2; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>2; + used = count*4; + while (count > 0) { + u_long data; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + *p++ = data ^ 0x80008000; + count--; + } + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ct_s16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + count = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data = le2be16(data); + *p++ = data; + *p++ = data; + count--; + } + *frameUsed += used*2; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>2; + used = count*4; + while (count > 0) { + u_long data; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + data = le2be16dbl(data); + *p++ = data; + count--; + } + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ct_u16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + count = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data = le2be16(data) ^ 0x8000; + *p++ = data; + *p++ = data; + } + *frameUsed += used*2; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>2; + used = count; + while (count > 0) { + u_long data; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + data = le2be16dbl(data) ^ 0x80008000; + *p++ = data; + count--; + } + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ctx_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8 + : dmasound_alaw2dma8; + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + u_char data = expand_data; + while (frameLeft) { + u_char c; + if (bal < 0) { + if (!userCount) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = table[c]; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_data = data; + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 2) { + u_char c; + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = table[c] << 8; + if (get_user(c, userPtr++)) + return -EFAULT; + data |= table[c]; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 2; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + u_char data = expand_data; + while (frameLeft) { + if (bal < 0) { + if (!userCount) + break; + if (get_user(data, userPtr++)) + return -EFAULT; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_data = data; + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 2) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 2; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + u_char data = expand_data; + while (frameLeft) { + if (bal < 0) { + if (!userCount) + break; + if (get_user(data, userPtr++)) + return -EFAULT; + data ^= 0x80; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_data = data; + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 2) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data ^= 0x8080; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 2; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_s16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_u16be(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data ^= 0x8000; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + data ^= 0x80008000; + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_s16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data = le2be16(data); + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + data = le2be16dbl(data); + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_u16le(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, ((u_short *)userPtr)++)) + return -EFAULT; + data = le2be16(data) ^ 0x8000; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, ((u_int *)userPtr)++)) + return -EFAULT; + data = le2be16dbl(data) ^ 0x80008000; + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static TRANS transTTNormal = { + .ct_ulaw = ata_ct_law, + .ct_alaw = ata_ct_law, + .ct_s8 = ata_ct_s8, + .ct_u8 = ata_ct_u8, +}; + +static TRANS transTTExpanding = { + .ct_ulaw = ata_ctx_law, + .ct_alaw = ata_ctx_law, + .ct_s8 = ata_ctx_s8, + .ct_u8 = ata_ctx_u8, +}; + +static TRANS transFalconNormal = { + .ct_ulaw = ata_ct_law, + .ct_alaw = ata_ct_law, + .ct_s8 = ata_ct_s8, + .ct_u8 = ata_ct_u8, + .ct_s16be = ata_ct_s16be, + .ct_u16be = ata_ct_u16be, + .ct_s16le = ata_ct_s16le, + .ct_u16le = ata_ct_u16le +}; + +static TRANS transFalconExpanding = { + .ct_ulaw = ata_ctx_law, + .ct_alaw = ata_ctx_law, + .ct_s8 = ata_ctx_s8, + .ct_u8 = ata_ctx_u8, + .ct_s16be = ata_ctx_s16be, + .ct_u16be = ata_ctx_u16be, + .ct_s16le = ata_ctx_s16le, + .ct_u16le = ata_ctx_u16le, +}; + + +/*** Low level stuff *********************************************************/ + + + +/* + * Atari (TT/Falcon) + */ + +static void *AtaAlloc(unsigned int size, int flags) +{ + return atari_stram_alloc(size, "dmasound"); +} + +static void AtaFree(void *obj, unsigned int size) +{ + atari_stram_free( obj ); +} + +static int __init AtaIrqInit(void) +{ + /* Set up timer A. Timer A + will receive a signal upon end of playing from the sound + hardware. Furthermore Timer A is able to count events + and will cause an interrupt after a programmed number + of events. So all we need to keep the music playing is + to provide the sound hardware with new data upon + an interrupt from timer A. */ + mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ + mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ + mfp.tim_ct_a = 8; /* Turn on event counting. */ + /* Register interrupt handler. */ + request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", + AtaInterrupt); + mfp.int_en_a |= 0x20; /* Turn interrupt on. */ + mfp.int_mk_a |= 0x20; + return 1; +} + +#ifdef MODULE +static void AtaIrqCleanUp(void) +{ + mfp.tim_ct_a = 0; /* stop timer */ + mfp.int_en_a &= ~0x20; /* turn interrupt off */ + free_irq(IRQ_MFP_TIMA, AtaInterrupt); +} +#endif /* MODULE */ + + +#define TONE_VOXWARE_TO_DB(v) \ + (((v) < 0) ? -12 : ((v) > 100) ? 12 : ((v) - 50) * 6 / 25) +#define TONE_DB_TO_VOXWARE(v) (((v) * 25 + ((v) > 0 ? 5 : -5)) / 6 + 50) + + +static int AtaSetBass(int bass) +{ + dmasound.bass = TONE_VOXWARE_TO_DB(bass); + atari_microwire_cmd(MW_LM1992_BASS(dmasound.bass)); + return TONE_DB_TO_VOXWARE(dmasound.bass); +} + + +static int AtaSetTreble(int treble) +{ + dmasound.treble = TONE_VOXWARE_TO_DB(treble); + atari_microwire_cmd(MW_LM1992_TREBLE(dmasound.treble)); + return TONE_DB_TO_VOXWARE(dmasound.treble); +} + + + +/* + * TT + */ + + +static void TTSilence(void) +{ + tt_dmasnd.ctrl = DMASND_CTRL_OFF; + atari_microwire_cmd(MW_LM1992_PSG_HIGH); /* mix in PSG signal 1:1 */ +} + + +static void TTInit(void) +{ + int mode, i, idx; + const int freq[4] = {50066, 25033, 12517, 6258}; + + /* search a frequency that fits into the allowed error range */ + + idx = -1; + for (i = 0; i < ARRAY_SIZE(freq); i++) + /* this isn't as much useful for a TT than for a Falcon, but + * then it doesn't hurt very much to implement it for a TT too. + */ + if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) < catchRadius) + idx = i; + if (idx > -1) { + dmasound.soft.speed = freq[idx]; + dmasound.trans_write = &transTTNormal; + } else + dmasound.trans_write = &transTTExpanding; + + TTSilence(); + dmasound.hard = dmasound.soft; + + if (dmasound.hard.speed > 50066) { + /* we would need to squeeze the sound, but we won't do that */ + dmasound.hard.speed = 50066; + mode = DMASND_MODE_50KHZ; + dmasound.trans_write = &transTTNormal; + } else if (dmasound.hard.speed > 25033) { + dmasound.hard.speed = 50066; + mode = DMASND_MODE_50KHZ; + } else if (dmasound.hard.speed > 12517) { + dmasound.hard.speed = 25033; + mode = DMASND_MODE_25KHZ; + } else if (dmasound.hard.speed > 6258) { + dmasound.hard.speed = 12517; + mode = DMASND_MODE_12KHZ; + } else { + dmasound.hard.speed = 6258; + mode = DMASND_MODE_6KHZ; + } + + tt_dmasnd.mode = (dmasound.hard.stereo ? + DMASND_MODE_STEREO : DMASND_MODE_MONO) | + DMASND_MODE_8BIT | mode; + + expand_bal = -dmasound.soft.speed; +} + + +static int TTSetFormat(int format) +{ + /* TT sound DMA supports only 8bit modes */ + + switch (format) { + case AFMT_QUERY: + return dmasound.soft.format; + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_S8: + case AFMT_U8: + break; + default: + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = 8; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = 8; + } + TTInit(); + + return format; +} + + +#define VOLUME_VOXWARE_TO_DB(v) \ + (((v) < 0) ? -40 : ((v) > 100) ? 0 : ((v) * 2) / 5 - 40) +#define VOLUME_DB_TO_VOXWARE(v) ((((v) + 40) * 5 + 1) / 2) + + +static int TTSetVolume(int volume) +{ + dmasound.volume_left = VOLUME_VOXWARE_TO_DB(volume & 0xff); + atari_microwire_cmd(MW_LM1992_BALLEFT(dmasound.volume_left)); + dmasound.volume_right = VOLUME_VOXWARE_TO_DB((volume & 0xff00) >> 8); + atari_microwire_cmd(MW_LM1992_BALRIGHT(dmasound.volume_right)); + return VOLUME_DB_TO_VOXWARE(dmasound.volume_left) | + (VOLUME_DB_TO_VOXWARE(dmasound.volume_right) << 8); +} + + +#define GAIN_VOXWARE_TO_DB(v) \ + (((v) < 0) ? -80 : ((v) > 100) ? 0 : ((v) * 4) / 5 - 80) +#define GAIN_DB_TO_VOXWARE(v) ((((v) + 80) * 5 + 1) / 4) + +static int TTSetGain(int gain) +{ + dmasound.gain = GAIN_VOXWARE_TO_DB(gain); + atari_microwire_cmd(MW_LM1992_VOLUME(dmasound.gain)); + return GAIN_DB_TO_VOXWARE(dmasound.gain); +} + + + +/* + * Falcon + */ + + +static void FalconSilence(void) +{ + /* stop playback, set sample rate 50kHz for PSG sound */ + tt_dmasnd.ctrl = DMASND_CTRL_OFF; + tt_dmasnd.mode = DMASND_MODE_50KHZ | DMASND_MODE_STEREO | DMASND_MODE_8BIT; + tt_dmasnd.int_div = 0; /* STE compatible divider */ + tt_dmasnd.int_ctrl = 0x0; + tt_dmasnd.cbar_src = 0x0000; /* no matrix inputs */ + tt_dmasnd.cbar_dst = 0x0000; /* no matrix outputs */ + tt_dmasnd.dac_src = 1; /* connect ADC to DAC, disconnect matrix */ + tt_dmasnd.adc_src = 3; /* ADC Input = PSG */ +} + + +static void FalconInit(void) +{ + int divider, i, idx; + const int freq[8] = {49170, 32780, 24585, 19668, 16390, 12292, 9834, 8195}; + + /* search a frequency that fits into the allowed error range */ + + idx = -1; + for (i = 0; i < ARRAY_SIZE(freq); i++) + /* if we will tolerate 3% error 8000Hz->8195Hz (2.38%) would + * be playable without expanding, but that now a kernel runtime + * option + */ + if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) < catchRadius) + idx = i; + if (idx > -1) { + dmasound.soft.speed = freq[idx]; + dmasound.trans_write = &transFalconNormal; + } else + dmasound.trans_write = &transFalconExpanding; + + FalconSilence(); + dmasound.hard = dmasound.soft; + + if (dmasound.hard.size == 16) { + /* the Falcon can play 16bit samples only in stereo */ + dmasound.hard.stereo = 1; + } + + if (dmasound.hard.speed > 49170) { + /* we would need to squeeze the sound, but we won't do that */ + dmasound.hard.speed = 49170; + divider = 1; + dmasound.trans_write = &transFalconNormal; + } else if (dmasound.hard.speed > 32780) { + dmasound.hard.speed = 49170; + divider = 1; + } else if (dmasound.hard.speed > 24585) { + dmasound.hard.speed = 32780; + divider = 2; + } else if (dmasound.hard.speed > 19668) { + dmasound.hard.speed = 24585; + divider = 3; + } else if (dmasound.hard.speed > 16390) { + dmasound.hard.speed = 19668; + divider = 4; + } else if (dmasound.hard.speed > 12292) { + dmasound.hard.speed = 16390; + divider = 5; + } else if (dmasound.hard.speed > 9834) { + dmasound.hard.speed = 12292; + divider = 7; + } else if (dmasound.hard.speed > 8195) { + dmasound.hard.speed = 9834; + divider = 9; + } else { + dmasound.hard.speed = 8195; + divider = 11; + } + tt_dmasnd.int_div = divider; + + /* Setup Falcon sound DMA for playback */ + tt_dmasnd.int_ctrl = 0x4; /* Timer A int at play end */ + tt_dmasnd.track_select = 0x0; /* play 1 track, track 1 */ + tt_dmasnd.cbar_src = 0x0001; /* DMA(25MHz) --> DAC */ + tt_dmasnd.cbar_dst = 0x0000; + tt_dmasnd.rec_track_select = 0; + tt_dmasnd.dac_src = 2; /* connect matrix to DAC */ + tt_dmasnd.adc_src = 0; /* ADC Input = Mic */ + + tt_dmasnd.mode = (dmasound.hard.stereo ? + DMASND_MODE_STEREO : DMASND_MODE_MONO) | + ((dmasound.hard.size == 8) ? + DMASND_MODE_8BIT : DMASND_MODE_16BIT) | + DMASND_MODE_6KHZ; + + expand_bal = -dmasound.soft.speed; +} + + +static int FalconSetFormat(int format) +{ + int size; + /* Falcon sound DMA supports 8bit and 16bit modes */ + + switch (format) { + case AFMT_QUERY: + return dmasound.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: /* :-) */ + size = 8; + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = size; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = dmasound.soft.size; + } + + FalconInit(); + + return format; +} + + +/* This is for the Falcon output *attenuation* in 1.5dB steps, + * i.e. output level from 0 to -22.5dB in -1.5dB steps. + */ +#define VOLUME_VOXWARE_TO_ATT(v) \ + ((v) < 0 ? 15 : (v) > 100 ? 0 : 15 - (v) * 3 / 20) +#define VOLUME_ATT_TO_VOXWARE(v) (100 - (v) * 20 / 3) + + +static int FalconSetVolume(int volume) +{ + dmasound.volume_left = VOLUME_VOXWARE_TO_ATT(volume & 0xff); + dmasound.volume_right = VOLUME_VOXWARE_TO_ATT((volume & 0xff00) >> 8); + tt_dmasnd.output_atten = dmasound.volume_left << 8 | dmasound.volume_right << 4; + return VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) | + VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8; +} + + +static void AtaPlayNextFrame(int index) +{ + char *start, *end; + + /* used by AtaPlay() if all doubts whether there really is something + * to be played are already wiped out. + */ + start = write_sq.buffers[write_sq.front]; + end = start+((write_sq.count == index) ? write_sq.rear_size + : write_sq.block_size); + /* end might not be a legal virtual address. */ + DMASNDSetEnd(virt_to_phys(end - 1) + 1); + DMASNDSetBase(virt_to_phys(start)); + /* Since only an even number of samples per frame can + be played, we might lose one byte here. (TO DO) */ + write_sq.front = (write_sq.front+1) % write_sq.max_count; + write_sq.active++; + tt_dmasnd.ctrl = DMASND_CTRL_ON | DMASND_CTRL_REPEAT; +} + + +static void AtaPlay(void) +{ + /* ++TeSche: Note that write_sq.active is no longer just a flag but + * holds the number of frames the DMA is currently programmed for + * instead, may be 0, 1 (currently being played) or 2 (pre-programmed). + * + * Changes done to write_sq.count and write_sq.active are a bit more + * subtle again so now I must admit I also prefer disabling the irq + * here rather than considering all possible situations. But the point + * is that disabling the irq doesn't have any bad influence on this + * version of the driver as we benefit from having pre-programmed the + * DMA wherever possible: There's no need to reload the DMA at the + * exact time of an interrupt but only at some time while the + * pre-programmed frame is playing! + */ + atari_disable_irq(IRQ_MFP_TIMA); + + if (write_sq.active == 2 || /* DMA is 'full' */ + write_sq.count <= 0) { /* nothing to do */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + + if (write_sq.active == 0) { + /* looks like there's nothing 'in' the DMA yet, so try + * to put two frames into it (at least one is available). + */ + if (write_sq.count == 1 && + write_sq.rear_size < write_sq.block_size && + !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + AtaPlayNextFrame(1); + if (write_sq.count == 1) { + /* no more frames */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + if (write_sq.count == 2 && + write_sq.rear_size < write_sq.block_size && + !write_sq.syncing) { + /* hmmm, there were two frames, but the second + * one is not yet filled and we're not syncing? + */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + AtaPlayNextFrame(2); + } else { + /* there's already a frame being played so we may only stuff + * one new into the DMA, but even if this may be the last + * frame existing the previous one is still on write_sq.count. + */ + if (write_sq.count == 2 && + write_sq.rear_size < write_sq.block_size && + !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + AtaPlayNextFrame(2); + } + atari_enable_irq(IRQ_MFP_TIMA); +} + + +static irqreturn_t AtaInterrupt(int irq, void *dummy, struct pt_regs *fp) +{ +#if 0 + /* ++TeSche: if you should want to test this... */ + static int cnt; + if (write_sq.active == 2) + if (++cnt == 10) { + /* simulate losing an interrupt */ + cnt = 0; + return IRQ_HANDLED; + } +#endif + spin_lock(&dmasound.lock); + if (write_sq_ignore_int && is_falcon) { + /* ++TeSche: Falcon only: ignore first irq because it comes + * immediately after starting a frame. after that, irqs come + * (almost) like on the TT. + */ + write_sq_ignore_int = 0; + return IRQ_HANDLED; + } + + if (!write_sq.active) { + /* playing was interrupted and sq_reset() has already cleared + * the sq variables, so better don't do anything here. + */ + WAKE_UP(write_sq.sync_queue); + return IRQ_HANDLED; + } + + /* Probably ;) one frame is finished. Well, in fact it may be that a + * pre-programmed one is also finished because there has been a long + * delay in interrupt delivery and we've completely lost one, but + * there's no way to detect such a situation. In such a case the last + * frame will be played more than once and the situation will recover + * as soon as the irq gets through. + */ + write_sq.count--; + write_sq.active--; + + if (!write_sq.active) { + tt_dmasnd.ctrl = DMASND_CTRL_OFF; + write_sq_ignore_int = 1; + } + + WAKE_UP(write_sq.action_queue); + /* At least one block of the queue is free now + so wake up a writing process blocked because + of a full queue. */ + + if ((write_sq.active != 1) || (write_sq.count != 1)) + /* We must be a bit carefully here: write_sq.count indicates the + * number of buffers used and not the number of frames to be + * played. If write_sq.count==1 and write_sq.active==1 that + * means the only remaining frame was already programmed + * earlier (and is currently running) so we mustn't call + * AtaPlay() here, otherwise we'll play one frame too much. + */ + AtaPlay(); + + if (!write_sq.active) WAKE_UP(write_sq.sync_queue); + /* We are not playing after AtaPlay(), so there + is nothing to play any more. Wake up a process + waiting for audio output to drain. */ + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} + + +/*** Mid level stuff *********************************************************/ + + +/* + * /dev/mixer abstraction + */ + +#define RECLEVEL_VOXWARE_TO_GAIN(v) \ + ((v) < 0 ? 0 : (v) > 100 ? 15 : (v) * 3 / 20) +#define RECLEVEL_GAIN_TO_VOXWARE(v) (((v) * 20 + 2) / 3) + + +static void __init TTMixerInit(void) +{ + atari_microwire_cmd(MW_LM1992_VOLUME(0)); + dmasound.volume_left = 0; + atari_microwire_cmd(MW_LM1992_BALLEFT(0)); + dmasound.volume_right = 0; + atari_microwire_cmd(MW_LM1992_BALRIGHT(0)); + atari_microwire_cmd(MW_LM1992_TREBLE(0)); + atari_microwire_cmd(MW_LM1992_BASS(0)); +} + +static void __init FalconMixerInit(void) +{ + dmasound.volume_left = (tt_dmasnd.output_atten & 0xf00) >> 8; + dmasound.volume_right = (tt_dmasnd.output_atten & 0xf0) >> 4; +} + +static int AtaMixerIoctl(u_int cmd, u_long arg) +{ + int data; + unsigned long flags; + switch (cmd) { + case SOUND_MIXER_READ_SPEAKER: + if (is_falcon || MACH_IS_TT) { + int porta; + spin_lock_irqsave(&dmasound.lock, flags); + sound_ym.rd_data_reg_sel = 14; + porta = sound_ym.rd_data_reg_sel; + spin_unlock_irqrestore(&dmasound.lock, flags); + return IOCTL_OUT(arg, porta & 0x40 ? 0 : 100); + } + break; + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_volume(data)); + case SOUND_MIXER_WRITE_SPEAKER: + if (is_falcon || MACH_IS_TT) { + int porta; + IOCTL_IN(arg, data); + spin_lock_irqsave(&dmasound.lock, flags); + sound_ym.rd_data_reg_sel = 14; + porta = (sound_ym.rd_data_reg_sel & ~0x40) | + (data < 50 ? 0x40 : 0); + sound_ym.wd_data = porta; + spin_unlock_irqrestore(&dmasound.lock, flags); + return IOCTL_OUT(arg, porta & 0x40 ? 0 : 100); + } + } + return -EINVAL; +} + + +static int TTMixerIoctl(u_int cmd, u_long arg) +{ + int data; + switch (cmd) { + case SOUND_MIXER_READ_RECMASK: + return IOCTL_OUT(arg, 0); + case SOUND_MIXER_READ_DEVMASK: + return IOCTL_OUT(arg, + SOUND_MASK_VOLUME | SOUND_MASK_TREBLE | SOUND_MASK_BASS | + (MACH_IS_TT ? SOUND_MASK_SPEAKER : 0)); + case SOUND_MIXER_READ_STEREODEVS: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME); + case SOUND_MIXER_READ_VOLUME: + return IOCTL_OUT(arg, + VOLUME_DB_TO_VOXWARE(dmasound.volume_left) | + (VOLUME_DB_TO_VOXWARE(dmasound.volume_right) << 8)); + case SOUND_MIXER_READ_BASS: + return IOCTL_OUT(arg, TONE_DB_TO_VOXWARE(dmasound.bass)); + case SOUND_MIXER_READ_TREBLE: + return IOCTL_OUT(arg, TONE_DB_TO_VOXWARE(dmasound.treble)); + case SOUND_MIXER_READ_OGAIN: + return IOCTL_OUT(arg, GAIN_DB_TO_VOXWARE(dmasound.gain)); + case SOUND_MIXER_WRITE_BASS: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_bass(data)); + case SOUND_MIXER_WRITE_TREBLE: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_treble(data)); + case SOUND_MIXER_WRITE_OGAIN: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_gain(data)); + } + return AtaMixerIoctl(cmd, arg); +} + +static int FalconMixerIoctl(u_int cmd, u_long arg) +{ + int data; + switch (cmd) { + case SOUND_MIXER_READ_RECMASK: + return IOCTL_OUT(arg, SOUND_MASK_MIC); + case SOUND_MIXER_READ_DEVMASK: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC | SOUND_MASK_SPEAKER); + case SOUND_MIXER_READ_STEREODEVS: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC); + case SOUND_MIXER_READ_VOLUME: + return IOCTL_OUT(arg, + VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) | + VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8); + case SOUND_MIXER_READ_CAPS: + return IOCTL_OUT(arg, SOUND_CAP_EXCL_INPUT); + case SOUND_MIXER_WRITE_MIC: + IOCTL_IN(arg, data); + tt_dmasnd.input_gain = + RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | + RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); + /* fall thru, return set value */ + case SOUND_MIXER_READ_MIC: + return IOCTL_OUT(arg, + RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | + RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain & 0xf) << 8); + } + return AtaMixerIoctl(cmd, arg); +} + +static int AtaWriteSqSetup(void) +{ + write_sq_ignore_int = 0; + return 0 ; +} + +static int AtaSqOpen(mode_t mode) +{ + write_sq_ignore_int = 1; + return 0 ; +} + +static int TTStateInfo(char *buffer, size_t space) +{ + int len = 0; + len += sprintf(buffer+len, "\tvol left %ddB [-40... 0]\n", + dmasound.volume_left); + len += sprintf(buffer+len, "\tvol right %ddB [-40... 0]\n", + dmasound.volume_right); + len += sprintf(buffer+len, "\tbass %ddB [-12...+12]\n", + dmasound.bass); + len += sprintf(buffer+len, "\ttreble %ddB [-12...+12]\n", + dmasound.treble); + if (len >= space) { + printk(KERN_ERR "dmasound_atari: overflowed state buffer alloc.\n") ; + len = space ; + } + return len; +} + +static int FalconStateInfo(char *buffer, size_t space) +{ + int len = 0; + len += sprintf(buffer+len, "\tvol left %ddB [-22.5 ... 0]\n", + dmasound.volume_left); + len += sprintf(buffer+len, "\tvol right %ddB [-22.5 ... 0]\n", + dmasound.volume_right); + if (len >= space) { + printk(KERN_ERR "dmasound_atari: overflowed state buffer alloc.\n") ; + len = space ; + } + return len; +} + + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard_falcon = { + .format = AFMT_S8, + .stereo = 0, + .size = 8, + .speed = 8195 +} ; + +static SETTINGS def_hard_tt = { + .format = AFMT_S8, + .stereo = 0, + .size = 8, + .speed = 12517 +} ; + +static SETTINGS def_soft = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static MACHINE machTT = { + .name = "Atari", + .name2 = "TT", + .owner = THIS_MODULE, + .dma_alloc = AtaAlloc, + .dma_free = AtaFree, + .irqinit = AtaIrqInit, +#ifdef MODULE + .irqcleanup = AtaIrqCleanUp, +#endif /* MODULE */ + .init = TTInit, + .silence = TTSilence, + .setFormat = TTSetFormat, + .setVolume = TTSetVolume, + .setBass = AtaSetBass, + .setTreble = AtaSetTreble, + .setGain = TTSetGain, + .play = AtaPlay, + .mixer_init = TTMixerInit, + .mixer_ioctl = TTMixerIoctl, + .write_sq_setup = AtaWriteSqSetup, + .sq_open = AtaSqOpen, + .state_info = TTStateInfo, + .min_dsp_speed = 6258, + .version = ((DMASOUND_ATARI_REVISION<<8) | DMASOUND_ATARI_EDITION), + .hardware_afmts = AFMT_S8, /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + +static MACHINE machFalcon = { + .name = "Atari", + .name2 = "FALCON", + .dma_alloc = AtaAlloc, + .dma_free = AtaFree, + .irqinit = AtaIrqInit, +#ifdef MODULE + .irqcleanup = AtaIrqCleanUp, +#endif /* MODULE */ + .init = FalconInit, + .silence = FalconSilence, + .setFormat = FalconSetFormat, + .setVolume = FalconSetVolume, + .setBass = AtaSetBass, + .setTreble = AtaSetTreble, + .play = AtaPlay, + .mixer_init = FalconMixerInit, + .mixer_ioctl = FalconMixerIoctl, + .write_sq_setup = AtaWriteSqSetup, + .sq_open = AtaSqOpen, + .state_info = FalconStateInfo, + .min_dsp_speed = 8195, + .version = ((DMASOUND_ATARI_REVISION<<8) | DMASOUND_ATARI_EDITION), + .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + + +/*** Config & Setup **********************************************************/ + + +static int __init dmasound_atari_init(void) +{ + if (MACH_IS_ATARI && ATARIHW_PRESENT(PCM_8BIT)) { + if (ATARIHW_PRESENT(CODEC)) { + dmasound.mach = machFalcon; + dmasound.mach.default_soft = def_soft ; + dmasound.mach.default_hard = def_hard_falcon ; + is_falcon = 1; + } else if (ATARIHW_PRESENT(MICROWIRE)) { + dmasound.mach = machTT; + dmasound.mach.default_soft = def_soft ; + dmasound.mach.default_hard = def_hard_tt ; + is_falcon = 0; + } else + return -ENODEV; + if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0) + return dmasound_init(); + else { + printk("DMA sound driver: Timer A interrupt already in use\n"); + return -EBUSY; + } + } + return -ENODEV; +} + +static void __exit dmasound_atari_cleanup(void) +{ + dmasound_deinit(); +} + +module_init(dmasound_atari_init); +module_exit(dmasound_atari_cleanup); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/dmasound/dmasound_awacs.c b/sound/oss/dmasound/dmasound_awacs.c new file mode 100644 index 0000000..5281b88 --- /dev/null +++ b/sound/oss/dmasound/dmasound_awacs.c @@ -0,0 +1,3176 @@ +/* + * linux/sound/oss/dmasound/dmasound_awacs.c + * + * PowerMac `AWACS' and `Burgundy' DMA Sound Driver + * with some limited support for DACA & Tumbler + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and + * history prior to 2001/01/26. + * + * 26/01/2001 ed 0.1 Iain Sandoe + * - added version info. + * - moved dbdma command buffer allocation to PMacXXXSqSetup() + * - fixed up beep dbdma cmd buffers + * + * 08/02/2001 [0.2] + * - make SNDCTL_DSP_GETFMTS return the correct info for the h/w + * - move soft format translations to a separate file + * - [0.3] make SNDCTL_DSP_GETCAPS return correct info. + * - [0.4] more informative machine name strings. + * - [0.5] + * - record changes. + * - made the default_hard/soft entries. + * 04/04/2001 [0.6] + * - minor correction to bit assignments in awacs_defs.h + * - incorporate mixer changes from 2.2.x back-port. + * - take out passthru as a rec input (it isn't). + * - make Input Gain slider work the 'right way up'. + * - try to make the mixer sliders more logical - so now the + * input selectors are just two-state (>50% == ON) and the + * Input Gain slider handles the rest of the gain issues. + * - try to pick slider representations that most closely match + * the actual use - e.g. IGain for input gain... + * - first stab at over/under-run detection. + * - minor cosmetic changes to IRQ identification. + * - fix bug where rates > max would be reported as supported. + * - first stab at over/under-run detection. + * - make use of i2c for mixer settings conditional on perch + * rather than cuda (some machines without perch have cuda). + * - fix bug where TX stops when dbdma status comes up "DEAD" + * so far only reported on PowerComputing clones ... but. + * - put in AWACS/Screamer register write timeouts. + * - part way to partitioning the init() stuff + * - first pass at 'tumbler' stuff (not support - just an attempt + * to allow the driver to load on new G4s). + * 01/02/2002 [0.7] - BenH + * - all sort of minor bits went in since the latest update, I + * bumped the version number for that reason + * + * 07/26/2002 [0.8] - BenH + * - More minor bits since last changelog (I should be more careful + * with those) + * - Support for snapper & better tumbler integration by Toby Sargeant + * - Headphone detect for scremer by Julien Blache + * - More tumbler fixed by Andreas Schwab + * 11/29/2003 [0.8.1] - Renzo Davoli (King Enzo) + * - Support for Snapper line in + * - snapper input resampling (for rates < 44100) + * - software line gain control + */ + +/* GENERAL FIXME/TODO: check that the assumptions about what is written to + mac-io is valid for DACA & Tumbler. + + This driver is in bad need of a rewrite. The dbdma code has to be split, + some proper device-tree parsing code has to be written, etc... +*/ + +#include <linux/types.h> +#include <linux/module.h> +#include <linux/config.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/soundcard.h> +#include <linux/adb.h> +#include <linux/nvram.h> +#include <linux/tty.h> +#include <linux/vt_kern.h> +#include <linux/spinlock.h> +#include <linux/kmod.h> +#include <linux/interrupt.h> +#include <linux/input.h> +#include <asm/semaphore.h> +#ifdef CONFIG_ADB_CUDA +#include <linux/cuda.h> +#endif +#ifdef CONFIG_ADB_PMU +#include <linux/pmu.h> +#endif + +#include <linux/i2c-dev.h> + +#include <asm/uaccess.h> +#include <asm/prom.h> +#include <asm/machdep.h> +#include <asm/io.h> +#include <asm/dbdma.h> +#include <asm/pmac_feature.h> +#include <asm/irq.h> +#include <asm/nvram.h> + +#include "awacs_defs.h" +#include "dmasound.h" +#include "tas3001c.h" +#include "tas3004.h" +#include "tas_common.h" + +#define DMASOUND_AWACS_REVISION 0 +#define DMASOUND_AWACS_EDITION 7 + +#define AWACS_SNAPPER 110 /* fake revision # for snapper */ +#define AWACS_BURGUNDY 100 /* fake revision # for burgundy */ +#define AWACS_TUMBLER 90 /* fake revision # for tumbler */ +#define AWACS_DACA 80 /* fake revision # for daca (ibook) */ +#define AWACS_AWACS 2 /* holding revision for AWACS */ +#define AWACS_SCREAMER 3 /* holding revision for Screamer */ +/* + * Interrupt numbers and addresses, & info obtained from the device tree. + */ +static int awacs_irq, awacs_tx_irq, awacs_rx_irq; +static volatile struct awacs_regs __iomem *awacs; +static volatile u32 __iomem *i2s; +static volatile struct dbdma_regs __iomem *awacs_txdma, *awacs_rxdma; +static int awacs_rate_index; +static int awacs_subframe; +static struct device_node* awacs_node; +static struct device_node* i2s_node; + +static char awacs_name[64]; +static int awacs_revision; +static int awacs_sleeping; +static DECLARE_MUTEX(dmasound_sem); + +static int sound_device_id; /* exists after iMac revA */ +static int hw_can_byteswap = 1 ; /* most pmac sound h/w can */ + +/* model info */ +/* To be replaced with better interaction with pmac_feature.c */ +static int is_pbook_3X00; +static int is_pbook_g3; + +/* expansion info */ +static int has_perch; +static int has_ziva; + +/* for earlier powerbooks which need fiddling with mac-io to enable + * cd etc. +*/ +static unsigned char __iomem *latch_base; +static unsigned char __iomem *macio_base; + +/* + * Space for the DBDMA command blocks. + */ +static void *awacs_tx_cmd_space; +static volatile struct dbdma_cmd *awacs_tx_cmds; +static int number_of_tx_cmd_buffers; + +static void *awacs_rx_cmd_space; +static volatile struct dbdma_cmd *awacs_rx_cmds; +static int number_of_rx_cmd_buffers; + +/* + * Cached values of AWACS registers (we can't read them). + * Except on the burgundy (and screamer). XXX + */ + +int awacs_reg[8]; +int awacs_reg1_save; + +/* tracking values for the mixer contents +*/ + +static int spk_vol; +static int line_vol; +static int passthru_vol; + +static int ip_gain; /* mic preamp settings */ +static int rec_lev = 0x4545 ; /* default CD gain 69 % */ +static int mic_lev; +static int cd_lev = 0x6363 ; /* 99 % */ +static int line_lev; + +static int hdp_connected; + +/* + * 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, +}; + +/* beep support */ +#define BEEP_SRATE 22050 /* 22050 Hz sample rate */ +#define BEEP_BUFLEN 512 +#define BEEP_VOLUME 15 /* 0 - 100 */ + +static int beep_vol = BEEP_VOLUME; +static int beep_playing; +static int awacs_beep_state; +static short *beep_buf; +static void *beep_dbdma_cmd_space; +static volatile struct dbdma_cmd *beep_dbdma_cmd; + +/* Burgundy functions */ +static void awacs_burgundy_wcw(unsigned addr,unsigned newval); +static unsigned awacs_burgundy_rcw(unsigned addr); +static void awacs_burgundy_write_volume(unsigned address, int volume); +static int awacs_burgundy_read_volume(unsigned address); +static void awacs_burgundy_write_mvolume(unsigned address, int volume); +static int awacs_burgundy_read_mvolume(unsigned address); + +/* we will allocate a single 'emergency' dbdma cmd block to use if the + tx status comes up "DEAD". This happens on some PowerComputing Pmac + clones, either owing to a bug in dbdma or some interaction between + IDE and sound. However, this measure would deal with DEAD status if + if appeared elsewhere. + + for the sake of memory efficiency we'll allocate this cmd as part of + the beep cmd stuff. +*/ + +static volatile struct dbdma_cmd *emergency_dbdma_cmd; + +#ifdef CONFIG_PMAC_PBOOK +/* + * Stuff for restoring after a sleep. + */ +static int awacs_sleep_notify(struct pmu_sleep_notifier *self, int when); +struct pmu_sleep_notifier awacs_sleep_notifier = { + awacs_sleep_notify, SLEEP_LEVEL_SOUND, +}; +#endif /* CONFIG_PMAC_PBOOK */ + +/* for (soft) sample rate translations */ +int expand_bal; /* Balance factor for expanding (not volume!) */ +int expand_read_bal; /* Balance factor for expanding reads (not volume!) */ + +/*** Low level stuff *********************************************************/ + +static void *PMacAlloc(unsigned int size, int flags); +static void PMacFree(void *ptr, unsigned int size); +static int PMacIrqInit(void); +#ifdef MODULE +static void PMacIrqCleanup(void); +#endif +static void PMacSilence(void); +static void PMacInit(void); +static int PMacSetFormat(int format); +static int PMacSetVolume(int volume); +static void PMacPlay(void); +static void PMacRecord(void); +static irqreturn_t pmac_awacs_tx_intr(int irq, void *devid, struct pt_regs *regs); +static irqreturn_t pmac_awacs_rx_intr(int irq, void *devid, struct pt_regs *regs); +static irqreturn_t pmac_awacs_intr(int irq, void *devid, struct pt_regs *regs); +static void awacs_write(int val); +static int awacs_get_volume(int reg, int lshift); +static int awacs_volume_setter(int volume, int n, int mute, int lshift); + + +/*** Mid level stuff **********************************************************/ + +static int PMacMixerIoctl(u_int cmd, u_long arg); +static int PMacWriteSqSetup(void); +static int PMacReadSqSetup(void); +static void PMacAbortRead(void); + +extern TRANS transAwacsNormal ; +extern TRANS transAwacsExpand ; +extern TRANS transAwacsNormalRead ; +extern TRANS transAwacsExpandRead ; + +extern int daca_init(void); +extern void daca_cleanup(void); +extern int daca_set_volume(uint left_vol, uint right_vol); +extern void daca_get_volume(uint * left_vol, uint *right_vol); +extern int daca_enter_sleep(void); +extern int daca_leave_sleep(void); + +#define TRY_LOCK() \ + if ((rc = down_interruptible(&dmasound_sem)) != 0) \ + return rc; +#define LOCK() down(&dmasound_sem); + +#define UNLOCK() up(&dmasound_sem); + +/* We use different versions that the ones provided in dmasound.h + * + * FIXME: Use different names ;) + */ +#undef IOCTL_IN +#undef IOCTL_OUT + +#define IOCTL_IN(arg, ret) \ + rc = get_user(ret, (int __user *)(arg)); \ + if (rc) break; +#define IOCTL_OUT(arg, ret) \ + ioctl_return2((int __user *)(arg), ret) + +static inline int ioctl_return2(int __user *addr, int value) +{ + return value < 0 ? value : put_user(value, addr); +} + + +/*** AE - TUMBLER / SNAPPER START ************************************************/ + + +int gpio_audio_reset, gpio_audio_reset_pol; +int gpio_amp_mute, gpio_amp_mute_pol; +int gpio_headphone_mute, gpio_headphone_mute_pol; +int gpio_headphone_detect, gpio_headphone_detect_pol; +int gpio_headphone_irq; + +int +setup_audio_gpio(const char *name, const char* compatible, int *gpio_addr, int* gpio_pol) +{ + struct device_node *np; + u32* pp; + + np = find_devices("gpio"); + if (!np) + return -ENODEV; + + np = np->child; + while(np != 0) { + if (name) { + char *property = get_property(np,"audio-gpio",NULL); + if (property != 0 && strcmp(property,name) == 0) + break; + } else if (compatible && device_is_compatible(np, compatible)) + break; + np = np->sibling; + } + if (!np) + return -ENODEV; + pp = (u32 *)get_property(np, "AAPL,address", NULL); + if (!pp) + return -ENODEV; + *gpio_addr = (*pp) & 0x0000ffff; + pp = (u32 *)get_property(np, "audio-gpio-active-state", NULL); + if (pp) + *gpio_pol = *pp; + else + *gpio_pol = 1; + if (np->n_intrs > 0) + return np->intrs[0].line; + + return 0; +} + +static inline void +write_audio_gpio(int gpio_addr, int data) +{ + if (!gpio_addr) + return; + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio_addr, data ? 0x05 : 0x04); +} + +static inline int +read_audio_gpio(int gpio_addr) +{ + if (!gpio_addr) + return 0; + return ((pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio_addr, 0) & 0x02) !=0); +} + +/* + * Headphone interrupt via GPIO (Tumbler, Snapper, DACA) + */ +static irqreturn_t +headphone_intr(int irq, void *devid, struct pt_regs *regs) +{ + unsigned long flags; + + spin_lock_irqsave(&dmasound.lock, flags); + if (read_audio_gpio(gpio_headphone_detect) == gpio_headphone_detect_pol) { + printk(KERN_INFO "Audio jack plugged, muting speakers.\n"); + write_audio_gpio(gpio_headphone_mute, !gpio_headphone_mute_pol); + write_audio_gpio(gpio_amp_mute, gpio_amp_mute_pol); + tas_output_device_change(sound_device_id,TAS_OUTPUT_HEADPHONES,0); + } else { + printk(KERN_INFO "Audio jack unplugged, enabling speakers.\n"); + write_audio_gpio(gpio_amp_mute, !gpio_amp_mute_pol); + write_audio_gpio(gpio_headphone_mute, gpio_headphone_mute_pol); + tas_output_device_change(sound_device_id,TAS_OUTPUT_INTERNAL_SPKR,0); + } + spin_unlock_irqrestore(&dmasound.lock, flags); + return IRQ_HANDLED; +} + + +/* Initialize tumbler */ + +static int +tas_dmasound_init(void) +{ + setup_audio_gpio( + "audio-hw-reset", + NULL, + &gpio_audio_reset, + &gpio_audio_reset_pol); + setup_audio_gpio( + "amp-mute", + NULL, + &gpio_amp_mute, + &gpio_amp_mute_pol); + setup_audio_gpio("headphone-mute", + NULL, + &gpio_headphone_mute, + &gpio_headphone_mute_pol); + gpio_headphone_irq = setup_audio_gpio( + "headphone-detect", + NULL, + &gpio_headphone_detect, + &gpio_headphone_detect_pol); + /* Fix some broken OF entries in desktop machines */ + if (!gpio_headphone_irq) + gpio_headphone_irq = setup_audio_gpio( + NULL, + "keywest-gpio15", + &gpio_headphone_detect, + &gpio_headphone_detect_pol); + + write_audio_gpio(gpio_audio_reset, gpio_audio_reset_pol); + msleep(100); + write_audio_gpio(gpio_audio_reset, !gpio_audio_reset_pol); + msleep(100); + if (gpio_headphone_irq) { + if (request_irq(gpio_headphone_irq,headphone_intr,0,"Headphone detect",NULL) < 0) { + printk(KERN_ERR "tumbler: Can't request headphone interrupt\n"); + gpio_headphone_irq = 0; + } else { + u8 val; + /* Activate headphone status interrupts */ + val = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio_headphone_detect, 0); + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio_headphone_detect, val | 0x80); + /* Trigger it */ + headphone_intr(0,NULL,NULL); + } + } + if (!gpio_headphone_irq) { + /* Some machine enter this case ? */ + printk(KERN_WARNING "tumbler: Headphone detect IRQ not found, enabling all outputs !\n"); + write_audio_gpio(gpio_amp_mute, !gpio_amp_mute_pol); + write_audio_gpio(gpio_headphone_mute, !gpio_headphone_mute_pol); + } + return 0; +} + + +static int +tas_dmasound_cleanup(void) +{ + if (gpio_headphone_irq) + free_irq(gpio_headphone_irq, NULL); + return 0; +} + +/* We don't support 48k yet */ +static int tas_freqs[1] = { 44100 } ; +static int tas_freqs_ok[1] = { 1 } ; + +/* don't know what to do really - just have to leave it where + * OF left things +*/ + +static int +tas_set_frame_rate(void) +{ + if (i2s) { + out_le32(i2s + (I2S_REG_SERIAL_FORMAT >> 2), 0x41190000); + out_le32(i2s + (I2S_REG_DATAWORD_SIZES >> 2), 0x02000200); + } + dmasound.hard.speed = 44100 ; + awacs_rate_index = 0 ; + return 44100 ; +} + +static int +tas_mixer_ioctl(u_int cmd, u_long arg) +{ + int __user *argp = (int __user *)arg; + int data; + int rc; + + rc=tas_device_ioctl(cmd, arg); + if (rc != -EINVAL) { + return rc; + } + + if ((cmd & ~0xff) == MIXER_WRITE(0) && + tas_supported_mixers() & (1<<(cmd & 0xff))) { + rc = get_user(data, argp); + if (rc<0) return rc; + tas_set_mixer_level(cmd & 0xff, data); + tas_get_mixer_level(cmd & 0xff, &data); + return ioctl_return2(argp, data); + } + if ((cmd & ~0xff) == MIXER_READ(0) && + tas_supported_mixers() & (1<<(cmd & 0xff))) { + tas_get_mixer_level(cmd & 0xff, &data); + return ioctl_return2(argp, data); + } + + switch(cmd) { + case SOUND_MIXER_READ_DEVMASK: + data = tas_supported_mixers() | SOUND_MASK_SPEAKER; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_STEREODEVS: + data = tas_stereo_mixers(); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_CAPS: + rc = IOCTL_OUT(arg, 0); + break; + case SOUND_MIXER_READ_RECMASK: + // XXX FIXME: find a way to check what is really available */ + data = SOUND_MASK_LINE | SOUND_MASK_MIC; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECSRC: + if (awacs_reg[0] & MASK_MUX_AUDIN) + data |= SOUND_MASK_LINE; + if (awacs_reg[0] & MASK_MUX_MIC) + data |= SOUND_MASK_MIC; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_RECSRC: + IOCTL_IN(arg, data); + data =0; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_SPEAKER: /* really bell volume */ + IOCTL_IN(arg, data); + beep_vol = data & 0xff; + /* fall through */ + case SOUND_MIXER_READ_SPEAKER: + rc = IOCTL_OUT(arg, (beep_vol<<8) | beep_vol); + break; + case SOUND_MIXER_OUTMASK: + case SOUND_MIXER_OUTSRC: + default: + rc = -EINVAL; + } + + return rc; +} + +static void __init +tas_init_frame_rates(unsigned int *prop, unsigned int l) +{ + int i ; + if (prop) { + for (i=0; i<1; i++) + tas_freqs_ok[i] = 0; + for (l /= sizeof(int); l > 0; --l) { + unsigned int r = *prop++; + /* Apple 'Fixed' format */ + if (r >= 0x10000) + r >>= 16; + for (i = 0; i < 1; ++i) { + if (r == tas_freqs[i]) { + tas_freqs_ok[i] = 1; + break; + } + } + } + } + /* else we assume that all the rates are available */ +} + + +/*** AE - TUMBLER / SNAPPER END ************************************************/ + + + +/*** Low level stuff *********************************************************/ + +/* + * PCI PowerMac, with AWACS, Screamer, Burgundy, DACA or Tumbler and DBDMA. + */ +static void *PMacAlloc(unsigned int size, int flags) +{ + return kmalloc(size, flags); +} + +static void PMacFree(void *ptr, unsigned int size) +{ + kfree(ptr); +} + +static int __init PMacIrqInit(void) +{ + if (awacs) + if (request_irq(awacs_irq, pmac_awacs_intr, 0, "Built-in Sound misc", NULL)) + return 0; + if (request_irq(awacs_tx_irq, pmac_awacs_tx_intr, 0, "Built-in Sound out", NULL) + || request_irq(awacs_rx_irq, pmac_awacs_rx_intr, 0, "Built-in Sound in", NULL)) + return 0; + return 1; +} + +#ifdef MODULE +static void PMacIrqCleanup(void) +{ + /* turn off input & output dma */ + DBDMA_DO_STOP(awacs_txdma); + DBDMA_DO_STOP(awacs_rxdma); + + if (awacs) + /* disable interrupts from awacs interface */ + out_le32(&awacs->control, in_le32(&awacs->control) & 0xfff); + + /* Switch off the sound clock */ + pmac_call_feature(PMAC_FTR_SOUND_CHIP_ENABLE, awacs_node, 0, 0); + /* Make sure proper bits are set on pismo & tipb */ + if ((machine_is_compatible("PowerBook3,1") || + machine_is_compatible("PowerBook3,2")) && awacs) { + awacs_reg[1] |= MASK_PAROUT0 | MASK_PAROUT1; + awacs_write(MASK_ADDR1 | awacs_reg[1]); + msleep(200); + } + if (awacs) + free_irq(awacs_irq, NULL); + free_irq(awacs_tx_irq, NULL); + free_irq(awacs_rx_irq, NULL); + + if (awacs) + iounmap(awacs); + if (i2s) + iounmap(i2s); + iounmap(awacs_txdma); + iounmap(awacs_rxdma); + + release_OF_resource(awacs_node, 0); + release_OF_resource(awacs_node, 1); + release_OF_resource(awacs_node, 2); + + if (awacs_tx_cmd_space) + kfree(awacs_tx_cmd_space); + if (awacs_rx_cmd_space) + kfree(awacs_rx_cmd_space); + if (beep_dbdma_cmd_space) + kfree(beep_dbdma_cmd_space); + if (beep_buf) + kfree(beep_buf); +#ifdef CONFIG_PMAC_PBOOK + pmu_unregister_sleep_notifier(&awacs_sleep_notifier); +#endif +} +#endif /* MODULE */ + +static void PMacSilence(void) +{ + /* turn off output dma */ + DBDMA_DO_STOP(awacs_txdma); +} + +/* don't know what to do really - just have to leave it where + * OF left things +*/ + +static int daca_set_frame_rate(void) +{ + if (i2s) { + out_le32(i2s + (I2S_REG_SERIAL_FORMAT >> 2), 0x41190000); + out_le32(i2s + (I2S_REG_DATAWORD_SIZES >> 2), 0x02000200); + } + dmasound.hard.speed = 44100 ; + awacs_rate_index = 0 ; + return 44100 ; +} + +static int awacs_freqs[8] = { + 44100, 29400, 22050, 17640, 14700, 11025, 8820, 7350 +}; +static int awacs_freqs_ok[8] = { 1, 1, 1, 1, 1, 1, 1, 1 }; + +static int +awacs_set_frame_rate(int desired, int catch_r) +{ + int tolerance, i = 8 ; + /* + * 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. + * N.B.: burgundy awacs only works at 44100 Hz. + */ + do { + tolerance = catch_r * awacs_freqs[--i] / 100; + if (awacs_freqs_ok[i] + && dmasound.soft.speed <= awacs_freqs[i] + tolerance) + break; + } while (i > 0); + dmasound.hard.speed = awacs_freqs[i]; + awacs_rate_index = i; + + out_le32(&awacs->control, MASK_IEPC | (i << 8) | 0x11 ); + awacs_reg[1] = (awacs_reg[1] & ~MASK_SAMPLERATE) | (i << 3); + awacs_write(awacs_reg[1] | MASK_ADDR1); + return dmasound.hard.speed; +} + +static int +burgundy_set_frame_rate(void) +{ + awacs_rate_index = 0 ; + awacs_reg[1] = (awacs_reg[1] & ~MASK_SAMPLERATE) ; + /* XXX disable error interrupt on burgundy for now */ + out_le32(&awacs->control, MASK_IEPC | 0 | 0x11 | MASK_IEE); + return 44100 ; +} + +static int +set_frame_rate(int desired, int catch_r) +{ + switch (awacs_revision) { + case AWACS_BURGUNDY: + dmasound.hard.speed = burgundy_set_frame_rate(); + break ; + case AWACS_TUMBLER: + case AWACS_SNAPPER: + dmasound.hard.speed = tas_set_frame_rate(); + break ; + case AWACS_DACA: + dmasound.hard.speed = + daca_set_frame_rate(); + break ; + default: + dmasound.hard.speed = awacs_set_frame_rate(desired, + catch_r); + break ; + } + return dmasound.hard.speed ; +} + +static void +awacs_recalibrate(void) +{ + /* Sorry for the horrible delays... I hope to get that improved + * by making the whole PM process asynchronous in a future version + */ + msleep(750); + awacs_reg[1] |= MASK_CMUTE | MASK_AMUTE; + awacs_write(awacs_reg[1] | MASK_RECALIBRATE | MASK_ADDR1); + msleep(1000); + awacs_write(awacs_reg[1] | MASK_ADDR1); +} + +static void PMacInit(void) +{ + int tolerance; + + switch (dmasound.soft.format) { + case AFMT_S16_LE: + case AFMT_U16_LE: + if (hw_can_byteswap) + dmasound.hard.format = AFMT_S16_LE; + else + dmasound.hard.format = AFMT_S16_BE; + break; + default: + dmasound.hard.format = AFMT_S16_BE; + break; + } + dmasound.hard.stereo = 1; + dmasound.hard.size = 16; + + /* set dmasound.hard.speed - on the basis of what we want (soft) + * and the tolerance we'll allow. + */ + set_frame_rate(dmasound.soft.speed, catchRadius) ; + + tolerance = (catchRadius * dmasound.hard.speed) / 100; + if (dmasound.soft.speed >= dmasound.hard.speed - tolerance) { + dmasound.trans_write = &transAwacsNormal; + dmasound.trans_read = &transAwacsNormalRead; + } else { + dmasound.trans_write = &transAwacsExpand; + dmasound.trans_read = &transAwacsExpandRead; + } + + if (awacs) { + if (hw_can_byteswap && (dmasound.hard.format == AFMT_S16_LE)) + out_le32(&awacs->byteswap, BS_VAL); + else + out_le32(&awacs->byteswap, 0); + } + + expand_bal = -dmasound.soft.speed; + expand_read_bal = -dmasound.soft.speed; +} + +static int PMacSetFormat(int format) +{ + int size; + int req_format = format; + + switch (format) { + case AFMT_QUERY: + return dmasound.soft.format; + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_U8: + case AFMT_S8: + size = 8; + break; + case AFMT_S16_LE: + if(!hw_can_byteswap) + format = AFMT_S16_BE; + case AFMT_S16_BE: + size = 16; + break; + case AFMT_U16_LE: + if(!hw_can_byteswap) + format = AFMT_U16_BE; + case AFMT_U16_BE: + size = 16; + break; + default: /* :-) */ + printk(KERN_ERR "dmasound: unknown format 0x%x, using AFMT_U8\n", + format); + size = 8; + format = AFMT_U8; + } + + if (req_format == format) { + dmasound.soft.format = format; + dmasound.soft.size = size; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = size; + } + } + + return format; +} + +#define AWACS_VOLUME_TO_MASK(x) (15 - ((((x) - 1) * 15) / 99)) +#define AWACS_MASK_TO_VOLUME(y) (100 - ((y) * 99 / 15)) + +static int awacs_get_volume(int reg, int lshift) +{ + int volume; + + volume = AWACS_MASK_TO_VOLUME((reg >> lshift) & 0xf); + volume |= AWACS_MASK_TO_VOLUME(reg & 0xf) << 8; + return volume; +} + +static int awacs_volume_setter(int volume, int n, int mute, int lshift) +{ + int r1, rn; + + if (mute && volume == 0) { + r1 = awacs_reg[1] | mute; + } else { + r1 = awacs_reg[1] & ~mute; + rn = awacs_reg[n] & ~(0xf | (0xf << lshift)); + rn |= ((AWACS_VOLUME_TO_MASK(volume & 0xff) & 0xf) << lshift); + rn |= AWACS_VOLUME_TO_MASK((volume >> 8) & 0xff) & 0xf; + awacs_reg[n] = rn; + awacs_write((n << 12) | rn); + volume = awacs_get_volume(rn, lshift); + } + if (r1 != awacs_reg[1]) { + awacs_reg[1] = r1; + awacs_write(r1 | MASK_ADDR1); + } + return volume; +} + +static int PMacSetVolume(int volume) +{ + printk(KERN_WARNING "Bogus call to PMacSetVolume !\n"); + return 0; +} + +static void awacs_setup_for_beep(int speed) +{ + out_le32(&awacs->control, + (in_le32(&awacs->control) & ~0x1f00) + | ((speed > 0 ? speed : awacs_rate_index) << 8)); + + if (hw_can_byteswap && (dmasound.hard.format == AFMT_S16_LE) && speed == -1) + out_le32(&awacs->byteswap, BS_VAL); + else + out_le32(&awacs->byteswap, 0); +} + +/* CHECK: how much of this *really* needs IRQs masked? */ +static void __PMacPlay(void) +{ + volatile struct dbdma_cmd *cp; + int next_frg, count; + + count = 300 ; /* > two cycles at the lowest sample rate */ + + /* what we want to send next */ + next_frg = (write_sq.front + write_sq.active) % write_sq.max_count; + + if (awacs_beep_state) { + /* sound takes precedence over beeps */ + /* stop the dma channel */ + out_le32(&awacs_txdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16); + while ( (in_le32(&awacs_txdma->status) & RUN) && count--) + udelay(1); + if (awacs) + awacs_setup_for_beep(-1); + out_le32(&awacs_txdma->cmdptr, + virt_to_bus(&(awacs_tx_cmds[next_frg]))); + + beep_playing = 0; + awacs_beep_state = 0; + } + /* this won't allow more than two frags to be in the output queue at + once. (or one, if the max frags is 2 - because count can't exceed + 2 in that case) + */ + while (write_sq.active < 2 && write_sq.active < write_sq.count) { + count = (write_sq.count == write_sq.active + 1) ? + write_sq.rear_size:write_sq.block_size ; + if (count < write_sq.block_size) { + if (!write_sq.syncing) /* last block not yet filled,*/ + break; /* and we're not syncing or POST-ed */ + else { + /* pretend the block is full to force a new + block to be started on the next write */ + write_sq.rear_size = write_sq.block_size ; + write_sq.syncing &= ~2 ; /* clear POST */ + } + } + cp = &awacs_tx_cmds[next_frg]; + st_le16(&cp->req_count, count); + st_le16(&cp->xfer_status, 0); + st_le16(&cp->command, OUTPUT_MORE + INTR_ALWAYS); + /* put a STOP at the end of the queue - but only if we have + space for it. This means that, if we under-run and we only + have two fragments, we might re-play sound from an existing + queued frag. I guess the solution to that is not to set two + frags if you are likely to under-run... + */ + if (write_sq.count < write_sq.max_count) { + if (++next_frg >= write_sq.max_count) + next_frg = 0 ; /* wrap */ + /* if we get here then we've underrun so we will stop*/ + st_le16(&awacs_tx_cmds[next_frg].command, DBDMA_STOP); + } + /* set the dbdma controller going, if it is not already */ + if (write_sq.active == 0) + out_le32(&awacs_txdma->cmdptr, virt_to_bus(cp)); + (void)in_le32(&awacs_txdma->status); + out_le32(&awacs_txdma->control, ((RUN|WAKE) << 16) + (RUN|WAKE)); + ++write_sq.active; + } +} + +static void PMacPlay(void) +{ + LOCK(); + if (!awacs_sleeping) { + unsigned long flags; + + spin_lock_irqsave(&dmasound.lock, flags); + __PMacPlay(); + spin_unlock_irqrestore(&dmasound.lock, flags); + } + UNLOCK(); +} + +static void PMacRecord(void) +{ + unsigned long flags; + + if (read_sq.active) + return; + + spin_lock_irqsave(&dmasound.lock, flags); + + /* This is all we have to do......Just start it up. + */ + out_le32(&awacs_rxdma->control, ((RUN|WAKE) << 16) + (RUN|WAKE)); + read_sq.active = 1; + + spin_unlock_irqrestore(&dmasound.lock, flags); +} + +/* if the TX status comes up "DEAD" - reported on some Power Computing machines + we need to re-start the dbdma - but from a different physical start address + and with a different transfer length. It would get very messy to do this + with the normal dbdma_cmd blocks - we would have to re-write the buffer start + addresses each time. So, we will keep a single dbdma_cmd block which can be + fiddled with. + When DEAD status is first reported the content of the faulted dbdma block is + copied into the emergency buffer and we note that the buffer is in use. + we then bump the start physical address by the amount that was successfully + output before it died. + On any subsequent DEAD result we just do the bump-ups (we know that we are + already using the emergency dbdma_cmd). + CHECK: this just tries to "do it". It is possible that we should abandon + xfers when the number of residual bytes gets below a certain value - I can + see that this might cause a loop-forever if too small a transfer causes + DEAD status. However this is a TODO for now - we'll see what gets reported. + When we get a successful transfer result with the emergency buffer we just + pretend that it completed using the original dmdma_cmd and carry on. The + 'next_cmd' field will already point back to the original loop of blocks. +*/ + +static irqreturn_t +pmac_awacs_tx_intr(int irq, void *devid, struct pt_regs *regs) +{ + int i = write_sq.front; + int stat; + int i_nowrap = write_sq.front; + volatile struct dbdma_cmd *cp; + /* != 0 when we are dealing with a DEAD xfer */ + static int emergency_in_use; + + spin_lock(&dmasound.lock); + while (write_sq.active > 0) { /* we expect to have done something*/ + if (emergency_in_use) /* we are dealing with DEAD xfer */ + cp = emergency_dbdma_cmd ; + else + cp = &awacs_tx_cmds[i]; + stat = ld_le16(&cp->xfer_status); + if (stat & DEAD) { + unsigned short req, res ; + unsigned int phy ; +#ifdef DEBUG_DMASOUND +printk("dmasound_pmac: tx-irq: xfer died - patching it up...\n") ; +#endif + /* to clear DEAD status we must first clear RUN + set it to quiescent to be on the safe side */ + (void)in_le32(&awacs_txdma->status); + out_le32(&awacs_txdma->control, + (RUN|PAUSE|FLUSH|WAKE) << 16); + write_sq.died++ ; + if (!emergency_in_use) { /* new problem */ + memcpy((void *)emergency_dbdma_cmd, (void *)cp, + sizeof(struct dbdma_cmd)); + emergency_in_use = 1; + cp = emergency_dbdma_cmd; + } + /* now bump the values to reflect the amount + we haven't yet shifted */ + req = ld_le16(&cp->req_count); + res = ld_le16(&cp->res_count); + phy = ld_le32(&cp->phy_addr); + phy += (req - res); + st_le16(&cp->req_count, res); + st_le16(&cp->res_count, 0); + st_le16(&cp->xfer_status, 0); + st_le32(&cp->phy_addr, phy); + st_le32(&cp->cmd_dep, virt_to_bus(&awacs_tx_cmds[(i+1)%write_sq.max_count])); + st_le16(&cp->command, OUTPUT_MORE | BR_ALWAYS | INTR_ALWAYS); + + /* point at our patched up command block */ + out_le32(&awacs_txdma->cmdptr, virt_to_bus(cp)); + /* we must re-start the controller */ + (void)in_le32(&awacs_txdma->status); + /* should complete clearing the DEAD status */ + out_le32(&awacs_txdma->control, + ((RUN|WAKE) << 16) + (RUN|WAKE)); + break; /* this block is still going */ + } + if ((stat & ACTIVE) == 0) + break; /* this frame is still going */ + if (emergency_in_use) + emergency_in_use = 0 ; /* done that */ + --write_sq.count; + --write_sq.active; + i_nowrap++; + if (++i >= write_sq.max_count) + i = 0; + } + + /* if we stopped and we were not sync-ing - then we under-ran */ + if( write_sq.syncing == 0 ){ + stat = in_le32(&awacs_txdma->status) ; + /* we hit the dbdma_stop */ + if( (stat & ACTIVE) == 0 ) write_sq.xruns++ ; + } + + /* if we used some data up then wake the writer to supply some more*/ + if (i_nowrap != write_sq.front) + WAKE_UP(write_sq.action_queue); + write_sq.front = i; + + /* but make sure we funnel what we've already got */\ + if (!awacs_sleeping) + __PMacPlay(); + + /* make the wake-on-empty conditional on syncing */ + if (!write_sq.active && (write_sq.syncing & 1)) + WAKE_UP(write_sq.sync_queue); /* any time we're empty */ + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} + + +static irqreturn_t +pmac_awacs_rx_intr(int irq, void *devid, struct pt_regs *regs) +{ + int stat ; + /* For some reason on my PowerBook G3, I get one interrupt + * when the interrupt vector is installed (like something is + * pending). This happens before the dbdma is initialized by + * us, so I just check the command pointer and if it is zero, + * just blow it off. + */ + if (in_le32(&awacs_rxdma->cmdptr) == 0) + return IRQ_HANDLED; + + /* We also want to blow 'em off when shutting down. + */ + if (read_sq.active == 0) + return IRQ_HANDLED; + + spin_lock(&dmasound.lock); + /* 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 ((stat=awacs_rx_cmds[read_sq.rear].xfer_status)) { + + /* if we got a "DEAD" status then just log it for now. + and try to restart dma. + TODO: figure out how best to fix it up + */ + if (stat & DEAD){ +#ifdef DEBUG_DMASOUND +printk("dmasound_pmac: rx-irq: DIED - attempting resurection\n"); +#endif + /* to clear DEAD status we must first clear RUN + set it to quiescent to be on the safe side */ + (void)in_le32(&awacs_txdma->status); + out_le32(&awacs_txdma->control, + (RUN|PAUSE|FLUSH|WAKE) << 16); + awacs_rx_cmds[read_sq.rear].xfer_status = 0; + awacs_rx_cmds[read_sq.rear].res_count = 0; + read_sq.died++ ; + (void)in_le32(&awacs_txdma->status); + /* re-start the same block */ + out_le32(&awacs_rxdma->cmdptr, + virt_to_bus(&awacs_rx_cmds[read_sq.rear])); + /* we must re-start the controller */ + (void)in_le32(&awacs_rxdma->status); + /* should complete clearing the DEAD status */ + out_le32(&awacs_rxdma->control, + ((RUN|WAKE) << 16) + (RUN|WAKE)); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; /* try this block again */ + } + /* Clear status and move on to next buffer. + */ + awacs_rx_cmds[read_sq.rear].xfer_status = 0; + 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++; + read_sq.xruns++ ; /* we overan */ + if (read_sq.front >= read_sq.max_active) + read_sq.front = 0; + } + } + + WAKE_UP(read_sq.action_queue); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} + + +static irqreturn_t +pmac_awacs_intr(int irq, void *devid, struct pt_regs *regs) +{ + int ctrl; + int status; + int r1; + + spin_lock(&dmasound.lock); + ctrl = in_le32(&awacs->control); + status = in_le32(&awacs->codec_stat); + + if (ctrl & MASK_PORTCHG) { + /* tested on Screamer, should work on others too */ + if (awacs_revision == AWACS_SCREAMER) { + if (((status & MASK_HDPCONN) >> 3) && (hdp_connected == 0)) { + hdp_connected = 1; + + r1 = awacs_reg[1] | MASK_SPKMUTE; + awacs_reg[1] = r1; + awacs_write(r1 | MASK_ADDR_MUTE); + } else if (((status & MASK_HDPCONN) >> 3 == 0) && (hdp_connected == 1)) { + hdp_connected = 0; + + r1 = awacs_reg[1] & ~MASK_SPKMUTE; + awacs_reg[1] = r1; + awacs_write(r1 | MASK_ADDR_MUTE); + } + } + } + if (ctrl & MASK_CNTLERR) { + int err = (in_le32(&awacs->codec_stat) & MASK_ERRCODE) >> 16; + /* CHECK: we just swallow burgundy errors at the moment..*/ + if (err != 0 && awacs_revision != AWACS_BURGUNDY) + printk(KERN_ERR "dmasound_pmac: error %x\n", err); + } + /* Writing 1s to the CNTLERR and PORTCHG bits clears them... */ + out_le32(&awacs->control, ctrl); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} + +static void +awacs_write(int val) +{ + int count = 300 ; + if (awacs_revision >= AWACS_DACA || !awacs) + return ; + + while ((in_le32(&awacs->codec_ctrl) & MASK_NEWECMD) && count--) + udelay(1) ; /* timeout is > 2 samples at lowest rate */ + out_le32(&awacs->codec_ctrl, val | (awacs_subframe << 22)); + (void)in_le32(&awacs->byteswap); +} + +/* this is called when the beep timer expires... it will be called even + if the beep has been overidden by other sound output. +*/ +static void awacs_nosound(unsigned long xx) +{ + unsigned long flags; + int count = 600 ; /* > four samples at lowest rate */ + + spin_lock_irqsave(&dmasound.lock, flags); + if (beep_playing) { + st_le16(&beep_dbdma_cmd->command, DBDMA_STOP); + out_le32(&awacs_txdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16); + while ((in_le32(&awacs_txdma->status) & RUN) && count--) + udelay(1); + if (awacs) + awacs_setup_for_beep(-1); + beep_playing = 0; + } + spin_unlock_irqrestore(&dmasound.lock, flags); +} + +/* + * We generate the beep with a single dbdma command that loops a buffer + * forever - without generating interrupts. + * + * So, to stop it you have to stop dma output as per awacs_nosound. + */ +static int awacs_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + unsigned long flags; + int beep_speed = 0; + int srate; + 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 (type != EV_SND) + return -1; + switch (code) { + case SND_BELL: + if (hz) + hz = 1000; + break; + case SND_TONE: + break; + default: + return -1; + } + + if (beep_buf == NULL) + return -1; + + /* quick-hack fix for DACA, Burgundy & Tumbler */ + + if (awacs_revision >= AWACS_DACA){ + srate = 44100 ; + } else { + for (i = 0; i < 8 && awacs_freqs[i] >= BEEP_SRATE; ++i) + if (awacs_freqs_ok[i]) + beep_speed = i; + srate = awacs_freqs[beep_speed]; + } + + if (hz <= srate / BEEP_BUFLEN || hz > srate / 2) { + /* cancel beep currently playing */ + awacs_nosound(0); + return 0; + } + + spin_lock_irqsave(&dmasound.lock, flags); + if (beep_playing || write_sq.active || beep_buf == NULL) { + spin_unlock_irqrestore(&dmasound.lock, flags); + return -1; /* too hard, sorry :-( */ + } + beep_playing = 1; + st_le16(&beep_dbdma_cmd->command, OUTPUT_MORE + BR_ALWAYS); + spin_unlock_irqrestore(&dmasound.lock, flags); + + if (hz == beep_hz_cache && beep_vol == 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_vol; + j = (j + f) & 0xffff; + } + beep_hz_cache = hz; + beep_volume_cache = beep_vol; + beep_nsamples_cache = nsamples; + } + + 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(&dmasound.lock, flags); + if (beep_playing) { /* i.e. haven't been terminated already */ + int count = 300 ; + out_le32(&awacs_txdma->control, (RUN|WAKE|FLUSH|PAUSE) << 16); + while ((in_le32(&awacs_txdma->status) & RUN) && count--) + udelay(1); /* timeout > 2 samples at lowest rate*/ + if (awacs) + awacs_setup_for_beep(beep_speed); + out_le32(&awacs_txdma->cmdptr, virt_to_bus(beep_dbdma_cmd)); + (void)in_le32(&awacs_txdma->status); + out_le32(&awacs_txdma->control, RUN | (RUN << 16)); + } + spin_unlock_irqrestore(&dmasound.lock, flags); + + return 0; +} + +/* used in init and for wake-up */ + +static void +load_awacs(void) +{ + awacs_write(awacs_reg[0] + MASK_ADDR0); + awacs_write(awacs_reg[1] + MASK_ADDR1); + awacs_write(awacs_reg[2] + MASK_ADDR2); + awacs_write(awacs_reg[4] + MASK_ADDR4); + + if (awacs_revision == AWACS_SCREAMER) { + awacs_write(awacs_reg[5] + MASK_ADDR5); + msleep(100); + awacs_write(awacs_reg[6] + MASK_ADDR6); + msleep(2); + awacs_write(awacs_reg[1] + MASK_ADDR1); + awacs_write(awacs_reg[7] + MASK_ADDR7); + } + if (awacs) { + if (hw_can_byteswap && (dmasound.hard.format == AFMT_S16_LE)) + out_le32(&awacs->byteswap, BS_VAL); + else + out_le32(&awacs->byteswap, 0); + } +} + +#ifdef CONFIG_PMAC_PBOOK +/* + * Save state when going to sleep, restore it afterwards. + */ +/* FIXME: sort out disabling/re-enabling of read stuff as well */ +static int awacs_sleep_notify(struct pmu_sleep_notifier *self, int when) +{ + unsigned long flags; + + switch (when) { + case PBOOK_SLEEP_NOW: + LOCK(); + awacs_sleeping = 1; + /* Tell the rest of the driver we are now going to sleep */ + mb(); + if (awacs_revision == AWACS_SCREAMER || + awacs_revision == AWACS_AWACS) { + awacs_reg1_save = awacs_reg[1]; + awacs_reg[1] |= MASK_AMUTE | MASK_CMUTE; + awacs_write(MASK_ADDR1 | awacs_reg[1]); + } + + PMacSilence(); + /* stop rx - if going - a bit of a daft user... but */ + out_le32(&awacs_rxdma->control, (RUN|WAKE|FLUSH << 16)); + /* deny interrupts */ + if (awacs) + disable_irq(awacs_irq); + disable_irq(awacs_tx_irq); + disable_irq(awacs_rx_irq); + /* Chip specific sleep code */ + switch (awacs_revision) { + case AWACS_TUMBLER: + case AWACS_SNAPPER: + write_audio_gpio(gpio_headphone_mute, gpio_headphone_mute_pol); + write_audio_gpio(gpio_amp_mute, gpio_amp_mute_pol); + tas_enter_sleep(); + write_audio_gpio(gpio_audio_reset, gpio_audio_reset_pol); + break ; + case AWACS_DACA: + daca_enter_sleep(); + break ; + case AWACS_BURGUNDY: + break ; + case AWACS_SCREAMER: + case AWACS_AWACS: + default: + out_le32(&awacs->control, 0x11) ; + break ; + } + /* Disable sound clock */ + pmac_call_feature(PMAC_FTR_SOUND_CHIP_ENABLE, awacs_node, 0, 0); + /* According to Darwin, we do that after turning off the sound + * chip clock. All this will have to be cleaned up once we properly + * parse the OF sound-objects + */ + if ((machine_is_compatible("PowerBook3,1") || + machine_is_compatible("PowerBook3,2")) && awacs) { + awacs_reg[1] |= MASK_PAROUT0 | MASK_PAROUT1; + awacs_write(MASK_ADDR1 | awacs_reg[1]); + msleep(200); + } + break; + case PBOOK_WAKE: + /* Enable sound clock */ + pmac_call_feature(PMAC_FTR_SOUND_CHIP_ENABLE, awacs_node, 0, 1); + if ((machine_is_compatible("PowerBook3,1") || + machine_is_compatible("PowerBook3,2")) && awacs) { + msleep(100); + awacs_reg[1] &= ~(MASK_PAROUT0 | MASK_PAROUT1); + awacs_write(MASK_ADDR1 | awacs_reg[1]); + msleep(300); + } else + msleep(1000); + /* restore settings */ + switch (awacs_revision) { + case AWACS_TUMBLER: + case AWACS_SNAPPER: + write_audio_gpio(gpio_headphone_mute, gpio_headphone_mute_pol); + write_audio_gpio(gpio_amp_mute, gpio_amp_mute_pol); + write_audio_gpio(gpio_audio_reset, gpio_audio_reset_pol); + msleep(100); + write_audio_gpio(gpio_audio_reset, !gpio_audio_reset_pol); + msleep(150); + tas_leave_sleep(); /* Stub for now */ + headphone_intr(0,NULL,NULL); + break; + case AWACS_DACA: + msleep(10); /* Check this !!! */ + daca_leave_sleep(); + break ; /* dont know how yet */ + case AWACS_BURGUNDY: + break ; + case AWACS_SCREAMER: + case AWACS_AWACS: + default: + load_awacs() ; + break ; + } + /* Recalibrate chip */ + if (awacs_revision == AWACS_SCREAMER && awacs) + awacs_recalibrate(); + /* Make sure dma is stopped */ + PMacSilence(); + if (awacs) + enable_irq(awacs_irq); + enable_irq(awacs_tx_irq); + enable_irq(awacs_rx_irq); + if (awacs) { + /* OK, allow ints back again */ + out_le32(&awacs->control, MASK_IEPC + | (awacs_rate_index << 8) | 0x11 + | (awacs_revision < AWACS_DACA ? MASK_IEE: 0)); + } + if (macio_base && is_pbook_g3) { + /* FIXME: should restore the setup we had...*/ + out_8(macio_base + 0x37, 3); + } else if (is_pbook_3X00) { + in_8(latch_base + 0x190); + } + /* Remove mute */ + if (awacs_revision == AWACS_SCREAMER || + awacs_revision == AWACS_AWACS) { + awacs_reg[1] = awacs_reg1_save; + awacs_write(MASK_ADDR1 | awacs_reg[1]); + } + awacs_sleeping = 0; + /* Resume pending sounds. */ + /* we don't try to restart input... */ + spin_lock_irqsave(&dmasound.lock, flags); + __PMacPlay(); + spin_unlock_irqrestore(&dmasound.lock, flags); + UNLOCK(); + } + return PBOOK_SLEEP_OK; +} +#endif /* CONFIG_PMAC_PBOOK */ + + +/* All the burgundy functions: */ + +/* Waits for busy flag to clear */ +inline static void +awacs_burgundy_busy_wait(void) +{ + int count = 50; /* > 2 samples at 44k1 */ + while ((in_le32(&awacs->codec_ctrl) & MASK_NEWECMD) && count--) + udelay(1) ; +} + +inline static void +awacs_burgundy_extend_wait(void) +{ + int count = 50 ; /* > 2 samples at 44k1 */ + while ((!(in_le32(&awacs->codec_stat) & MASK_EXTEND)) && count--) + udelay(1) ; + count = 50; + while ((in_le32(&awacs->codec_stat) & MASK_EXTEND) && count--) + udelay(1); +} + +static void +awacs_burgundy_wcw(unsigned addr, unsigned val) +{ + out_le32(&awacs->codec_ctrl, addr + 0x200c00 + (val & 0xff)); + awacs_burgundy_busy_wait(); + out_le32(&awacs->codec_ctrl, addr + 0x200d00 +((val>>8) & 0xff)); + awacs_burgundy_busy_wait(); + out_le32(&awacs->codec_ctrl, addr + 0x200e00 +((val>>16) & 0xff)); + awacs_burgundy_busy_wait(); + out_le32(&awacs->codec_ctrl, addr + 0x200f00 +((val>>24) & 0xff)); + awacs_burgundy_busy_wait(); +} + +static unsigned +awacs_burgundy_rcw(unsigned addr) +{ + unsigned val = 0; + unsigned long flags; + + /* should have timeouts here */ + spin_lock_irqsave(&dmasound.lock, flags); + + out_le32(&awacs->codec_ctrl, addr + 0x100000); + awacs_burgundy_busy_wait(); + awacs_burgundy_extend_wait(); + val += (in_le32(&awacs->codec_stat) >> 4) & 0xff; + + out_le32(&awacs->codec_ctrl, addr + 0x100100); + awacs_burgundy_busy_wait(); + awacs_burgundy_extend_wait(); + val += ((in_le32(&awacs->codec_stat)>>4) & 0xff) <<8; + + out_le32(&awacs->codec_ctrl, addr + 0x100200); + awacs_burgundy_busy_wait(); + awacs_burgundy_extend_wait(); + val += ((in_le32(&awacs->codec_stat)>>4) & 0xff) <<16; + + out_le32(&awacs->codec_ctrl, addr + 0x100300); + awacs_burgundy_busy_wait(); + awacs_burgundy_extend_wait(); + val += ((in_le32(&awacs->codec_stat)>>4) & 0xff) <<24; + + spin_unlock_irqrestore(&dmasound.lock, flags); + + return val; +} + + +static void +awacs_burgundy_wcb(unsigned addr, unsigned val) +{ + out_le32(&awacs->codec_ctrl, addr + 0x300000 + (val & 0xff)); + awacs_burgundy_busy_wait(); +} + +static unsigned +awacs_burgundy_rcb(unsigned addr) +{ + unsigned val = 0; + unsigned long flags; + + /* should have timeouts here */ + spin_lock_irqsave(&dmasound.lock, flags); + + out_le32(&awacs->codec_ctrl, addr + 0x100000); + awacs_burgundy_busy_wait(); + awacs_burgundy_extend_wait(); + val += (in_le32(&awacs->codec_stat) >> 4) & 0xff; + + spin_unlock_irqrestore(&dmasound.lock, flags); + + return val; +} + +static int +awacs_burgundy_check(void) +{ + /* Checks to see the chip is alive and kicking */ + int error = in_le32(&awacs->codec_ctrl) & MASK_ERRCODE; + + return error == 0xf0000; +} + +static int +awacs_burgundy_init(void) +{ + if (awacs_burgundy_check()) { + printk(KERN_WARNING "dmasound_pmac: burgundy not working :-(\n"); + return 1; + } + + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_OUTPUTENABLES, + DEF_BURGUNDY_OUTPUTENABLES); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + DEF_BURGUNDY_MORE_OUTPUTENABLES); + awacs_burgundy_wcw(MASK_ADDR_BURGUNDY_OUTPUTSELECTS, + DEF_BURGUNDY_OUTPUTSELECTS); + + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_INPSEL21, + DEF_BURGUNDY_INPSEL21); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_INPSEL3, + DEF_BURGUNDY_INPSEL3); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_GAINCD, + DEF_BURGUNDY_GAINCD); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_GAINLINE, + DEF_BURGUNDY_GAINLINE); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_GAINMIC, + DEF_BURGUNDY_GAINMIC); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_GAINMODEM, + DEF_BURGUNDY_GAINMODEM); + + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_ATTENSPEAKER, + DEF_BURGUNDY_ATTENSPEAKER); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_ATTENLINEOUT, + DEF_BURGUNDY_ATTENLINEOUT); + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_ATTENHP, + DEF_BURGUNDY_ATTENHP); + + awacs_burgundy_wcw(MASK_ADDR_BURGUNDY_MASTER_VOLUME, + DEF_BURGUNDY_MASTER_VOLUME); + awacs_burgundy_wcw(MASK_ADDR_BURGUNDY_VOLCD, + DEF_BURGUNDY_VOLCD); + awacs_burgundy_wcw(MASK_ADDR_BURGUNDY_VOLLINE, + DEF_BURGUNDY_VOLLINE); + awacs_burgundy_wcw(MASK_ADDR_BURGUNDY_VOLMIC, + DEF_BURGUNDY_VOLMIC); + return 0; +} + +static void +awacs_burgundy_write_volume(unsigned address, int volume) +{ + int hardvolume,lvolume,rvolume; + + lvolume = (volume & 0xff) ? (volume & 0xff) + 155 : 0; + rvolume = ((volume >>8)&0xff) ? ((volume >> 8)&0xff ) + 155 : 0; + + hardvolume = lvolume + (rvolume << 16); + + awacs_burgundy_wcw(address, hardvolume); +} + +static int +awacs_burgundy_read_volume(unsigned address) +{ + int softvolume,wvolume; + + wvolume = awacs_burgundy_rcw(address); + + softvolume = (wvolume & 0xff) - 155; + softvolume += (((wvolume >> 16) & 0xff) - 155)<<8; + + return softvolume > 0 ? softvolume : 0; +} + +static int +awacs_burgundy_read_mvolume(unsigned address) +{ + int lvolume,rvolume,wvolume; + + wvolume = awacs_burgundy_rcw(address); + + wvolume &= 0xffff; + + rvolume = (wvolume & 0xff) - 155; + lvolume = ((wvolume & 0xff00)>>8) - 155; + + return lvolume + (rvolume << 8); +} + +static void +awacs_burgundy_write_mvolume(unsigned address, int volume) +{ + int lvolume,rvolume,hardvolume; + + lvolume = (volume &0xff) ? (volume & 0xff) + 155 :0; + rvolume = ((volume >>8) & 0xff) ? (volume >> 8) + 155 :0; + + hardvolume = lvolume + (rvolume << 8); + hardvolume += (hardvolume << 16); + + awacs_burgundy_wcw(address, hardvolume); +} + +/* End burgundy functions */ + +/* Set up output volumes on machines with the 'perch/whisper' extension card. + * this has an SGS i2c chip (7433) which is accessed using the cuda. + * + * TODO: split this out and make use of the other parts of the SGS chip to + * do Bass, Treble etc. + */ + +static void +awacs_enable_amp(int spkr_vol) +{ +#ifdef CONFIG_ADB_CUDA + struct adb_request req; + + if (sys_ctrler != SYS_CTRLER_CUDA) + return; + + /* turn on headphones */ + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, + 0x8a, 4, 0); + while (!req.complete) cuda_poll(); + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, + 0x8a, 6, 0); + while (!req.complete) cuda_poll(); + + /* turn on speaker */ + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, + 0x8a, 3, (100 - (spkr_vol & 0xff)) * 32 / 100); + while (!req.complete) cuda_poll(); + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, + 0x8a, 5, (100 - ((spkr_vol >> 8) & 0xff)) * 32 / 100); + while (!req.complete) cuda_poll(); + + cuda_request(&req, NULL, 5, CUDA_PACKET, + CUDA_GET_SET_IIC, 0x8a, 1, 0x29); + while (!req.complete) cuda_poll(); +#endif /* CONFIG_ADB_CUDA */ +} + + +/*** Mid level stuff *********************************************************/ + + +/* + * /dev/mixer abstraction + */ + +static void do_line_lev(int data) +{ + line_lev = data ; + awacs_reg[0] &= ~MASK_MUX_AUDIN; + if ((data & 0xff) >= 50) + awacs_reg[0] |= MASK_MUX_AUDIN; + awacs_write(MASK_ADDR0 | awacs_reg[0]); +} + +static void do_ip_gain(int data) +{ + ip_gain = data ; + data &= 0xff; + awacs_reg[0] &= ~MASK_GAINLINE; + if (awacs_revision == AWACS_SCREAMER) { + awacs_reg[6] &= ~MASK_MIC_BOOST ; + if (data >= 33) { + awacs_reg[0] |= MASK_GAINLINE; + if( data >= 66) + awacs_reg[6] |= MASK_MIC_BOOST ; + } + awacs_write(MASK_ADDR6 | awacs_reg[6]) ; + } else { + if (data >= 50) + awacs_reg[0] |= MASK_GAINLINE; + } + awacs_write(MASK_ADDR0 | awacs_reg[0]); +} + +static void do_mic_lev(int data) +{ + mic_lev = data ; + data &= 0xff; + awacs_reg[0] &= ~MASK_MUX_MIC; + if (data >= 50) + awacs_reg[0] |= MASK_MUX_MIC; + awacs_write(MASK_ADDR0 | awacs_reg[0]); +} + +static void do_cd_lev(int data) +{ + cd_lev = data ; + awacs_reg[0] &= ~MASK_MUX_CD; + if ((data & 0xff) >= 50) + awacs_reg[0] |= MASK_MUX_CD; + awacs_write(MASK_ADDR0 | awacs_reg[0]); +} + +static void do_rec_lev(int data) +{ + int left, right ; + rec_lev = data ; + /* need to fudge this to use the volume setter routine */ + left = 100 - (data & 0xff) ; if( left < 0 ) left = 0 ; + right = 100 - ((data >> 8) & 0xff) ; if( right < 0 ) right = 0 ; + left |= (right << 8 ); + left = awacs_volume_setter(left, 0, 0, 4); +} + +static void do_passthru_vol(int data) +{ + passthru_vol = data ; + awacs_reg[1] &= ~MASK_LOOPTHRU; + if (awacs_revision == AWACS_SCREAMER) { + if( data ) { /* switch it on for non-zero */ + awacs_reg[1] |= MASK_LOOPTHRU; + awacs_write(MASK_ADDR1 | awacs_reg[1]); + } + data = awacs_volume_setter(data, 5, 0, 6) ; + } else { + if ((data & 0xff) >= 50) + awacs_reg[1] |= MASK_LOOPTHRU; + awacs_write(MASK_ADDR1 | awacs_reg[1]); + data = (awacs_reg[1] & MASK_LOOPTHRU)? 100: 0; + } +} + +static int awacs_mixer_ioctl(u_int cmd, u_long arg) +{ + int data; + int rc; + + switch (cmd) { + case SOUND_MIXER_READ_CAPS: + /* say we will allow multiple inputs? prob. wrong + so I'm switching it to single */ + return IOCTL_OUT(arg, 1); + case SOUND_MIXER_READ_DEVMASK: + data = SOUND_MASK_VOLUME | SOUND_MASK_SPEAKER + | SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD + | SOUND_MASK_IGAIN | SOUND_MASK_RECLEV + | SOUND_MASK_ALTPCM + | SOUND_MASK_MONITOR; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECMASK: + data = SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECSRC: + data = 0; + if (awacs_reg[0] & MASK_MUX_AUDIN) + data |= SOUND_MASK_LINE; + if (awacs_reg[0] & MASK_MUX_MIC) + data |= SOUND_MASK_MIC; + if (awacs_reg[0] & MASK_MUX_CD) + data |= SOUND_MASK_CD; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_RECSRC: + IOCTL_IN(arg, data); + data &= (SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD); + awacs_reg[0] &= ~(MASK_MUX_CD | MASK_MUX_MIC + | MASK_MUX_AUDIN); + if (data & SOUND_MASK_LINE) + awacs_reg[0] |= MASK_MUX_AUDIN; + if (data & SOUND_MASK_MIC) + awacs_reg[0] |= MASK_MUX_MIC; + if (data & SOUND_MASK_CD) + awacs_reg[0] |= MASK_MUX_CD; + awacs_write(awacs_reg[0] | MASK_ADDR0); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_STEREODEVS: + data = SOUND_MASK_VOLUME | SOUND_MASK_SPEAKER| SOUND_MASK_RECLEV ; + if (awacs_revision == AWACS_SCREAMER) + data |= SOUND_MASK_MONITOR ; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + line_vol = data ; + awacs_volume_setter(data, 2, 0, 6); + /* fall through */ + case SOUND_MIXER_READ_VOLUME: + rc = IOCTL_OUT(arg, line_vol); + break; + case SOUND_MIXER_WRITE_SPEAKER: + IOCTL_IN(arg, data); + spk_vol = data ; + if (has_perch) + awacs_enable_amp(data); + else + (void)awacs_volume_setter(data, 4, MASK_CMUTE, 6); + /* fall though */ + case SOUND_MIXER_READ_SPEAKER: + rc = IOCTL_OUT(arg, spk_vol); + break; + case SOUND_MIXER_WRITE_ALTPCM: /* really bell volume */ + IOCTL_IN(arg, data); + beep_vol = data & 0xff; + /* fall through */ + case SOUND_MIXER_READ_ALTPCM: + rc = IOCTL_OUT(arg, beep_vol); + break; + case SOUND_MIXER_WRITE_LINE: + IOCTL_IN(arg, data); + do_line_lev(data) ; + /* fall through */ + case SOUND_MIXER_READ_LINE: + rc = IOCTL_OUT(arg, line_lev); + break; + case SOUND_MIXER_WRITE_IGAIN: + IOCTL_IN(arg, data); + do_ip_gain(data) ; + /* fall through */ + case SOUND_MIXER_READ_IGAIN: + rc = IOCTL_OUT(arg, ip_gain); + break; + case SOUND_MIXER_WRITE_MIC: + IOCTL_IN(arg, data); + do_mic_lev(data); + /* fall through */ + case SOUND_MIXER_READ_MIC: + rc = IOCTL_OUT(arg, mic_lev); + break; + case SOUND_MIXER_WRITE_CD: + IOCTL_IN(arg, data); + do_cd_lev(data); + /* fall through */ + case SOUND_MIXER_READ_CD: + rc = IOCTL_OUT(arg, cd_lev); + break; + case SOUND_MIXER_WRITE_RECLEV: + IOCTL_IN(arg, data); + do_rec_lev(data) ; + /* fall through */ + case SOUND_MIXER_READ_RECLEV: + rc = IOCTL_OUT(arg, rec_lev); + break; + case MIXER_WRITE(SOUND_MIXER_MONITOR): + IOCTL_IN(arg, data); + do_passthru_vol(data) ; + /* fall through */ + case MIXER_READ(SOUND_MIXER_MONITOR): + rc = IOCTL_OUT(arg, passthru_vol); + break; + default: + rc = -EINVAL; + } + + return rc; +} + +static void awacs_mixer_init(void) +{ + awacs_volume_setter(line_vol, 2, 0, 6); + if (has_perch) + awacs_enable_amp(spk_vol); + else + (void)awacs_volume_setter(spk_vol, 4, MASK_CMUTE, 6); + do_line_lev(line_lev) ; + do_ip_gain(ip_gain) ; + do_mic_lev(mic_lev) ; + do_cd_lev(cd_lev) ; + do_rec_lev(rec_lev) ; + do_passthru_vol(passthru_vol) ; +} + +static int burgundy_mixer_ioctl(u_int cmd, u_long arg) +{ + int data; + int rc; + + /* We are, we are, we are... Burgundy or better */ + switch(cmd) { + case SOUND_MIXER_READ_DEVMASK: + data = SOUND_MASK_VOLUME | SOUND_MASK_CD | + SOUND_MASK_LINE | SOUND_MASK_MIC | + SOUND_MASK_SPEAKER | SOUND_MASK_ALTPCM; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECMASK: + data = SOUND_MASK_LINE | SOUND_MASK_MIC + | SOUND_MASK_CD; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECSRC: + data = 0; + if (awacs_reg[0] & MASK_MUX_AUDIN) + data |= SOUND_MASK_LINE; + if (awacs_reg[0] & MASK_MUX_MIC) + data |= SOUND_MASK_MIC; + if (awacs_reg[0] & MASK_MUX_CD) + data |= SOUND_MASK_CD; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_RECSRC: + IOCTL_IN(arg, data); + data &= (SOUND_MASK_LINE + | SOUND_MASK_MIC | SOUND_MASK_CD); + awacs_reg[0] &= ~(MASK_MUX_CD | MASK_MUX_MIC + | MASK_MUX_AUDIN); + if (data & SOUND_MASK_LINE) + awacs_reg[0] |= MASK_MUX_AUDIN; + if (data & SOUND_MASK_MIC) + awacs_reg[0] |= MASK_MUX_MIC; + if (data & SOUND_MASK_CD) + awacs_reg[0] |= MASK_MUX_CD; + awacs_write(awacs_reg[0] | MASK_ADDR0); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_STEREODEVS: + data = SOUND_MASK_VOLUME | SOUND_MASK_SPEAKER + | SOUND_MASK_RECLEV | SOUND_MASK_CD + | SOUND_MASK_LINE; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_CAPS: + rc = IOCTL_OUT(arg, 0); + break; + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + awacs_burgundy_write_mvolume(MASK_ADDR_BURGUNDY_MASTER_VOLUME, data); + /* Fall through */ + case SOUND_MIXER_READ_VOLUME: + rc = IOCTL_OUT(arg, awacs_burgundy_read_mvolume(MASK_ADDR_BURGUNDY_MASTER_VOLUME)); + break; + case SOUND_MIXER_WRITE_SPEAKER: + IOCTL_IN(arg, data); + if (!(data & 0xff)) { + /* Mute the left speaker */ + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + awacs_burgundy_rcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES) & ~0x2); + } else { + /* Unmute the left speaker */ + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + awacs_burgundy_rcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES) | 0x2); + } + if (!(data & 0xff00)) { + /* Mute the right speaker */ + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + awacs_burgundy_rcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES) & ~0x4); + } else { + /* Unmute the right speaker */ + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + awacs_burgundy_rcb(MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES) | 0x4); + } + + data = (((data&0xff)*16)/100 > 0xf ? 0xf : + (((data&0xff)*16)/100)) + + ((((data>>8)*16)/100 > 0xf ? 0xf : + ((((data>>8)*16)/100)))<<4); + + awacs_burgundy_wcb(MASK_ADDR_BURGUNDY_ATTENSPEAKER, ~data); + /* Fall through */ + case SOUND_MIXER_READ_SPEAKER: + data = awacs_burgundy_rcb(MASK_ADDR_BURGUNDY_ATTENSPEAKER); + data = (((data & 0xf)*100)/16) + ((((data>>4)*100)/16)<<8); + rc = IOCTL_OUT(arg, (~data) & 0x0000ffff); + break; + case SOUND_MIXER_WRITE_ALTPCM: /* really bell volume */ + IOCTL_IN(arg, data); + beep_vol = data & 0xff; + /* fall through */ + case SOUND_MIXER_READ_ALTPCM: + rc = IOCTL_OUT(arg, beep_vol); + break; + case SOUND_MIXER_WRITE_LINE: + IOCTL_IN(arg, data); + awacs_burgundy_write_volume(MASK_ADDR_BURGUNDY_VOLLINE, data); + + /* fall through */ + case SOUND_MIXER_READ_LINE: + data = awacs_burgundy_read_volume(MASK_ADDR_BURGUNDY_VOLLINE); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_MIC: + IOCTL_IN(arg, data); + /* Mic is mono device */ + data = (data << 8) + (data << 24); + awacs_burgundy_write_volume(MASK_ADDR_BURGUNDY_VOLMIC, data); + /* fall through */ + case SOUND_MIXER_READ_MIC: + data = awacs_burgundy_read_volume(MASK_ADDR_BURGUNDY_VOLMIC); + data <<= 24; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_CD: + IOCTL_IN(arg, data); + awacs_burgundy_write_volume(MASK_ADDR_BURGUNDY_VOLCD, data); + /* fall through */ + case SOUND_MIXER_READ_CD: + data = awacs_burgundy_read_volume(MASK_ADDR_BURGUNDY_VOLCD); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_RECLEV: + IOCTL_IN(arg, data); + data = awacs_volume_setter(data, 0, 0, 4); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECLEV: + data = awacs_get_volume(awacs_reg[0], 4); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_OUTMASK: + case SOUND_MIXER_OUTSRC: + default: + rc = -EINVAL; + } + + return rc; +} + +static int daca_mixer_ioctl(u_int cmd, u_long arg) +{ + int data; + int rc; + + /* And the DACA's no genius either! */ + + switch(cmd) { + case SOUND_MIXER_READ_DEVMASK: + data = SOUND_MASK_VOLUME; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECMASK: + data = 0; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_RECSRC: + data = 0; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_WRITE_RECSRC: + IOCTL_IN(arg, data); + data =0; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_STEREODEVS: + data = SOUND_MASK_VOLUME; + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_READ_CAPS: + rc = IOCTL_OUT(arg, 0); + break; + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + daca_set_volume(data, data); + /* Fall through */ + case SOUND_MIXER_READ_VOLUME: + daca_get_volume(& data, &data); + rc = IOCTL_OUT(arg, data); + break; + case SOUND_MIXER_OUTMASK: + case SOUND_MIXER_OUTSRC: + default: + rc = -EINVAL; + } + return rc; +} + +static int PMacMixerIoctl(u_int cmd, u_long arg) +{ + int rc; + + /* Different IOCTLS for burgundy and, eventually, DACA & Tumbler */ + + TRY_LOCK(); + + switch (awacs_revision){ + case AWACS_BURGUNDY: + rc = burgundy_mixer_ioctl(cmd, arg); + break ; + case AWACS_DACA: + rc = daca_mixer_ioctl(cmd, arg); + break; + case AWACS_TUMBLER: + case AWACS_SNAPPER: + rc = tas_mixer_ioctl(cmd, arg); + break ; + default: /* ;-)) */ + rc = awacs_mixer_ioctl(cmd, arg); + } + + UNLOCK(); + + return rc; +} + +static void PMacMixerInit(void) +{ + switch (awacs_revision) { + case AWACS_TUMBLER: + printk("AE-Init tumbler mixer\n"); + break ; + case AWACS_SNAPPER: + printk("AE-Init snapper mixer\n"); + break ; + case AWACS_DACA: + case AWACS_BURGUNDY: + break ; /* don't know yet */ + case AWACS_AWACS: + case AWACS_SCREAMER: + default: + awacs_mixer_init() ; + break ; + } +} + +/* Write/Read sq setup functions: + Check to see if we have enough (or any) dbdma cmd buffers for the + user's fragment settings. If not, allocate some. If this fails we will + point at the beep buffer - as an emergency provision - to stop dma tromping + on some random bit of memory (if someone lets it go anyway). + The command buffers are then set up to point to the fragment buffers + (allocated elsewhere). We need n+1 commands the last of which holds + a NOP + loop to start. +*/ + +static int PMacWriteSqSetup(void) +{ + int i, count = 600 ; + volatile struct dbdma_cmd *cp; + + LOCK(); + + /* stop the controller from doing any output - if it isn't already. + it _should_ be before this is called anyway */ + + out_le32(&awacs_txdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16); + while ((in_le32(&awacs_txdma->status) & RUN) && count--) + udelay(1); +#ifdef DEBUG_DMASOUND +if (count <= 0) + printk("dmasound_pmac: write sq setup: timeout waiting for dma to stop\n"); +#endif + + if ((write_sq.max_count + 1) > number_of_tx_cmd_buffers) { + if (awacs_tx_cmd_space) + kfree(awacs_tx_cmd_space); + number_of_tx_cmd_buffers = 0; + + /* we need nbufs + 1 (for the loop) and we should request + 1 + again because the DBDMA_ALIGN might pull the start up by up + to sizeof(struct dbdma_cmd) - 4. + */ + + awacs_tx_cmd_space = kmalloc + ((write_sq.max_count + 1 + 1) * sizeof(struct dbdma_cmd), + GFP_KERNEL); + if (awacs_tx_cmd_space == NULL) { + /* don't leave it dangling - nasty but better than a + random address */ + out_le32(&awacs_txdma->cmdptr, virt_to_bus(beep_dbdma_cmd)); + printk(KERN_ERR + "dmasound_pmac: can't allocate dbdma cmd buffers" + ", driver disabled\n"); + UNLOCK(); + return -ENOMEM; + } + awacs_tx_cmds = (volatile struct dbdma_cmd *) + DBDMA_ALIGN(awacs_tx_cmd_space); + number_of_tx_cmd_buffers = write_sq.max_count + 1; + } + + cp = awacs_tx_cmds; + memset((void *)cp, 0, (write_sq.max_count+1) * sizeof(struct dbdma_cmd)); + for (i = 0; i < write_sq.max_count; ++i, ++cp) { + st_le32(&cp->phy_addr, virt_to_bus(write_sq.buffers[i])); + } + st_le16(&cp->command, DBDMA_NOP + BR_ALWAYS); + st_le32(&cp->cmd_dep, virt_to_bus(awacs_tx_cmds)); + /* point the controller at the command stack - ready to go */ + out_le32(&awacs_txdma->cmdptr, virt_to_bus(awacs_tx_cmds)); + UNLOCK(); + return 0; +} + +static int PMacReadSqSetup(void) +{ + int i, count = 600; + volatile struct dbdma_cmd *cp; + + LOCK(); + + /* stop the controller from doing any input - if it isn't already. + it _should_ be before this is called anyway */ + + out_le32(&awacs_rxdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16); + while ((in_le32(&awacs_rxdma->status) & RUN) && count--) + udelay(1); +#ifdef DEBUG_DMASOUND +if (count <= 0) + printk("dmasound_pmac: read sq setup: timeout waiting for dma to stop\n"); +#endif + + if ((read_sq.max_count+1) > number_of_rx_cmd_buffers ) { + if (awacs_rx_cmd_space) + kfree(awacs_rx_cmd_space); + number_of_rx_cmd_buffers = 0; + + /* we need nbufs + 1 (for the loop) and we should request + 1 again + because the DBDMA_ALIGN might pull the start up by up to + sizeof(struct dbdma_cmd) - 4 (assuming kmalloc aligns 32 bits). + */ + + awacs_rx_cmd_space = kmalloc + ((read_sq.max_count + 1 + 1) * sizeof(struct dbdma_cmd), + GFP_KERNEL); + if (awacs_rx_cmd_space == NULL) { + /* don't leave it dangling - nasty but better than a + random address */ + out_le32(&awacs_rxdma->cmdptr, virt_to_bus(beep_dbdma_cmd)); + printk(KERN_ERR + "dmasound_pmac: can't allocate dbdma cmd buffers" + ", driver disabled\n"); + UNLOCK(); + return -ENOMEM; + } + awacs_rx_cmds = (volatile struct dbdma_cmd *) + DBDMA_ALIGN(awacs_rx_cmd_space); + number_of_rx_cmd_buffers = read_sq.max_count + 1 ; + } + cp = awacs_rx_cmds; + memset((void *)cp, 0, (read_sq.max_count+1) * sizeof(struct dbdma_cmd)); + + /* Set dma buffers up in a loop */ + for (i = 0; i < read_sq.max_count; i++,cp++) { + st_le32(&cp->phy_addr, virt_to_bus(read_sq.buffers[i])); + st_le16(&cp->command, INPUT_MORE + INTR_ALWAYS); + st_le16(&cp->req_count, read_sq.block_size); + st_le16(&cp->xfer_status, 0); + } + + /* The next two lines make the thing loop around. + */ + st_le16(&cp->command, DBDMA_NOP + BR_ALWAYS); + st_le32(&cp->cmd_dep, virt_to_bus(awacs_rx_cmds)); + /* point the controller at the command stack - ready to go */ + out_le32(&awacs_rxdma->cmdptr, virt_to_bus(awacs_rx_cmds)); + + UNLOCK(); + return 0; +} + +/* TODO: this needs work to guarantee that when it returns DMA has stopped + but in a more elegant way than is done here.... +*/ + +static void PMacAbortRead(void) +{ + int i; + volatile struct dbdma_cmd *cp; + + LOCK(); + /* give it a chance to update the output and provide the IRQ + that is expected. + */ + + out_le32(&awacs_rxdma->control, ((FLUSH) << 16) + FLUSH ); + + cp = awacs_rx_cmds; + for (i = 0; i < read_sq.max_count; i++,cp++) + st_le16(&cp->command, DBDMA_STOP); + /* + * We should probably wait for the thing to stop before we + * release the memory. + */ + + msleep(100) ; /* give it a (small) chance to act */ + + /* apply the sledgehammer approach - just stop it now */ + + out_le32(&awacs_rxdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16); + UNLOCK(); +} + +extern char *get_afmt_string(int); +static int PMacStateInfo(char *b, size_t sp) +{ + int i, len = 0; + len = sprintf(b,"HW rates: "); + switch (awacs_revision){ + case AWACS_DACA: + case AWACS_BURGUNDY: + len += sprintf(b,"44100 ") ; + break ; + case AWACS_TUMBLER: + case AWACS_SNAPPER: + for (i=0; i<1; i++){ + if (tas_freqs_ok[i]) + len += sprintf(b+len,"%d ", tas_freqs[i]) ; + } + break ; + case AWACS_AWACS: + case AWACS_SCREAMER: + default: + for (i=0; i<8; i++){ + if (awacs_freqs_ok[i]) + len += sprintf(b+len,"%d ", awacs_freqs[i]) ; + } + break ; + } + len += sprintf(b+len,"s/sec\n") ; + if (len < sp) { + len += sprintf(b+len,"HW AFMTS: "); + i = AFMT_U16_BE ; + while (i) { + if (i & dmasound.mach.hardware_afmts) + len += sprintf(b+len,"%s ", + get_afmt_string(i & dmasound.mach.hardware_afmts)); + i >>= 1 ; + } + len += sprintf(b+len,"\n") ; + } + return len ; +} + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard = { + .format = AFMT_S16_BE, + .stereo = 1, + .size = 16, + .speed = 44100 +} ; + +static SETTINGS def_soft = { + .format = AFMT_S16_BE, + .stereo = 1, + .size = 16, + .speed = 44100 +} ; + +static MACHINE machPMac = { + .name = awacs_name, + .name2 = "PowerMac Built-in Sound", + .owner = THIS_MODULE, + .dma_alloc = PMacAlloc, + .dma_free = PMacFree, + .irqinit = PMacIrqInit, +#ifdef MODULE + .irqcleanup = PMacIrqCleanup, +#endif /* MODULE */ + .init = PMacInit, + .silence = PMacSilence, + .setFormat = PMacSetFormat, + .setVolume = PMacSetVolume, + .play = PMacPlay, + .record = NULL, /* default to no record */ + .mixer_init = PMacMixerInit, + .mixer_ioctl = PMacMixerIoctl, + .write_sq_setup = PMacWriteSqSetup, + .read_sq_setup = PMacReadSqSetup, + .state_info = PMacStateInfo, + .abort_read = PMacAbortRead, + .min_dsp_speed = 7350, + .max_dsp_speed = 44100, + .version = ((DMASOUND_AWACS_REVISION<<8) + DMASOUND_AWACS_EDITION) +}; + + +/*** Config & Setup **********************************************************/ + +/* Check for pmac models that we care about in terms of special actions. +*/ + +void __init +set_model(void) +{ + /* portables/lap-tops */ + + if (machine_is_compatible("AAPL,3400/2400") || + machine_is_compatible("AAPL,3500")) { + is_pbook_3X00 = 1 ; + } + if (machine_is_compatible("PowerBook1,1") || /* lombard */ + machine_is_compatible("AAPL,PowerBook1998")){ /* wallstreet */ + is_pbook_g3 = 1 ; + return ; + } +} + +/* Get the OF node that tells us about the registers, interrupts etc. to use + for sound IO. + + On most machines the sound IO OF node is the 'davbus' node. On newer pmacs + with DACA (& Tumbler) the node to use is i2s-a. On much older machines i.e. + before 9500 there is no davbus node and we have to use the 'awacs' property. + + In the latter case we signal this by setting the codec value - so that the + code that looks for chip properties knows how to go about it. +*/ + +static struct device_node* __init +get_snd_io_node(void) +{ + struct device_node *np = NULL; + + /* set up awacs_node for early OF which doesn't have a full set of + * properties on davbus + */ + + awacs_node = find_devices("awacs"); + if (awacs_node) + awacs_revision = AWACS_AWACS; + + /* powermac models after 9500 (other than those which use DACA or + * Tumbler) have a node called "davbus". + */ + np = find_devices("davbus"); + /* + * if we didn't find a davbus device, try 'i2s-a' since + * this seems to be what iBooks (& Tumbler) have. + */ + if (np == NULL) + np = i2s_node = find_devices("i2s-a"); + + /* if we didn't find this - perhaps we are on an early model + * which _only_ has an 'awacs' node + */ + if (np == NULL && awacs_node) + np = awacs_node ; + + /* if we failed all these return null - this will cause the + * driver to give up... + */ + return np ; +} + +/* Get the OF node that contains the info about the sound chip, inputs s-rates + etc. + This node does not exist (or contains much reduced info) on earlier machines + we have to deduce the info other ways for these. +*/ + +static struct device_node* __init +get_snd_info_node(struct device_node *io) +{ + struct device_node *info; + + info = find_devices("sound"); + while (info && info->parent != io) + info = info->next; + return info; +} + +/* Find out what type of codec we have. +*/ + +static int __init +get_codec_type(struct device_node *info) +{ + /* already set if pre-davbus model and info will be NULL */ + int codec = awacs_revision ; + + if (info) { + /* must do awacs first to allow screamer to overide it */ + if (device_is_compatible(info, "awacs")) + codec = AWACS_AWACS ; + if (device_is_compatible(info, "screamer")) + codec = AWACS_SCREAMER; + if (device_is_compatible(info, "burgundy")) + codec = AWACS_BURGUNDY ; + if (device_is_compatible(info, "daca")) + codec = AWACS_DACA; + if (device_is_compatible(info, "tumbler")) + codec = AWACS_TUMBLER; + if (device_is_compatible(info, "snapper")) + codec = AWACS_SNAPPER; + } + return codec ; +} + +/* find out what type, if any, of expansion card we have +*/ +static void __init +get_expansion_type(void) +{ + if (find_devices("perch") != NULL) + has_perch = 1; + + if (find_devices("pb-ziva-pc") != NULL) + has_ziva = 1; + /* need to work out how we deal with iMac SRS module */ +} + +/* set up frame rates. + * I suspect that these routines don't quite go about it the right way: + * - where there is more than one rate - I think that the first property + * value is the number of rates. + * TODO: check some more device trees and modify accordingly + * Set dmasound.mach.max_dsp_rate on the basis of these routines. +*/ + +static void __init +awacs_init_frame_rates(unsigned int *prop, unsigned int l) +{ + int i ; + if (prop) { + for (i=0; i<8; i++) + awacs_freqs_ok[i] = 0 ; + for (l /= sizeof(int); l > 0; --l) { + unsigned int r = *prop++; + /* Apple 'Fixed' format */ + if (r >= 0x10000) + r >>= 16; + for (i = 0; i < 8; ++i) { + if (r == awacs_freqs[i]) { + awacs_freqs_ok[i] = 1; + break; + } + } + } + } + /* else we assume that all the rates are available */ +} + +static void __init +burgundy_init_frame_rates(unsigned int *prop, unsigned int l) +{ + int temp[9] ; + int i = 0 ; + if (prop) { + for (l /= sizeof(int); l > 0; --l) { + unsigned int r = *prop++; + /* Apple 'Fixed' format */ + if (r >= 0x10000) + r >>= 16; + temp[i] = r ; + i++ ; if(i>=9) i=8; + } + } +#ifdef DEBUG_DMASOUND +if (i > 1){ + int j; + printk("dmasound_pmac: burgundy with multiple frame rates\n"); + for(j=0; j<i; j++) + printk("%d ", temp[j]) ; + printk("\n") ; +} +#endif +} + +static void __init +daca_init_frame_rates(unsigned int *prop, unsigned int l) +{ + int temp[9] ; + int i = 0 ; + if (prop) { + for (l /= sizeof(int); l > 0; --l) { + unsigned int r = *prop++; + /* Apple 'Fixed' format */ + if (r >= 0x10000) + r >>= 16; + temp[i] = r ; + i++ ; if(i>=9) i=8; + + } + } +#ifdef DEBUG_DMASOUND +if (i > 1){ + int j; + printk("dmasound_pmac: DACA with multiple frame rates\n"); + for(j=0; j<i; j++) + printk("%d ", temp[j]) ; + printk("\n") ; +} +#endif +} + +static void __init +init_frame_rates(unsigned int *prop, unsigned int l) +{ + switch (awacs_revision) { + case AWACS_TUMBLER: + case AWACS_SNAPPER: + tas_init_frame_rates(prop, l); + break ; + case AWACS_DACA: + daca_init_frame_rates(prop, l); + break ; + case AWACS_BURGUNDY: + burgundy_init_frame_rates(prop, l); + break ; + default: + awacs_init_frame_rates(prop, l); + break ; + } +} + +/* find things/machines that can't do mac-io byteswap +*/ + +static void __init +set_hw_byteswap(struct device_node *io) +{ + struct device_node *mio ; + unsigned int kl = 0 ; + + /* if seems that Keylargo can't byte-swap */ + + for (mio = io->parent; mio ; mio = mio->parent) { + if (strcmp(mio->name, "mac-io") == 0) { + if (device_is_compatible(mio, "Keylargo")) + kl = 1; + break; + } + } + hw_can_byteswap = !kl; +} + +/* Allocate the resources necessary for beep generation. This cannot be (quite) + done statically (yet) because we cannot do virt_to_bus() on static vars when + the code is loaded as a module. + + for the sake of saving the possibility that two allocations will incur the + overhead of two pull-ups in DBDMA_ALIGN() we allocate the 'emergency' dmdma + command here as well... even tho' it is not part of the beep process. +*/ + +int32_t +__init setup_beep(void) +{ + /* Initialize beep stuff */ + /* want one cmd buffer for beeps, and a second one for emergencies + - i.e. dbdma error conditions. + ask for three to allow for pull up in DBDMA_ALIGN(). + */ + beep_dbdma_cmd_space = + kmalloc((2 + 1) * sizeof(struct dbdma_cmd), GFP_KERNEL); + if(beep_dbdma_cmd_space == NULL) { + printk(KERN_ERR "dmasound_pmac: no beep dbdma cmd space\n") ; + return -ENOMEM ; + } + beep_dbdma_cmd = (volatile struct dbdma_cmd *) + DBDMA_ALIGN(beep_dbdma_cmd_space); + /* set up emergency dbdma cmd */ + emergency_dbdma_cmd = beep_dbdma_cmd+1 ; + beep_buf = (short *) kmalloc(BEEP_BUFLEN * 4, GFP_KERNEL); + if (beep_buf == NULL) { + printk(KERN_ERR "dmasound_pmac: no memory for beep buffer\n"); + if( beep_dbdma_cmd_space ) kfree(beep_dbdma_cmd_space) ; + return -ENOMEM ; + } + return 0 ; +} + +static struct input_dev awacs_beep_dev = { + .evbit = { BIT(EV_SND) }, + .sndbit = { BIT(SND_BELL) | BIT(SND_TONE) }, + .event = awacs_beep_event, + .name = "dmasound beeper", + .phys = "macio/input0", /* what the heck is this?? */ + .id = { + .bustype = BUS_HOST, + }, +}; + +int __init dmasound_awacs_init(void) +{ + struct device_node *io = NULL, *info = NULL; + int vol, res; + + if (_machine != _MACH_Pmac) + return -ENODEV; + + awacs_subframe = 0; + awacs_revision = 0; + hw_can_byteswap = 1 ; /* most can */ + + /* look for models we need to handle specially */ + set_model() ; + + /* find the OF node that tells us about the dbdma stuff + */ + io = get_snd_io_node(); + if (io == NULL) { +#ifdef DEBUG_DMASOUND +printk("dmasound_pmac: couldn't find sound io OF node\n"); +#endif + return -ENODEV ; + } + + /* find the OF node that tells us about the sound sub-system + * this doesn't exist on pre-davbus machines (earlier than 9500) + */ + if (awacs_revision != AWACS_AWACS) { /* set for pre-davbus */ + info = get_snd_info_node(io) ; + if (info == NULL){ +#ifdef DEBUG_DMASOUND +printk("dmasound_pmac: couldn't find 'sound' OF node\n"); +#endif + return -ENODEV ; + } + } + + awacs_revision = get_codec_type(info) ; + if (awacs_revision == 0) { +#ifdef DEBUG_DMASOUND +printk("dmasound_pmac: couldn't find a Codec we can handle\n"); +#endif + return -ENODEV ; /* we don't know this type of h/w */ + } + + /* set up perch, ziva, SRS or whatever else we have as sound + * expansion. + */ + get_expansion_type(); + + /* we've now got enough information to make up the audio topology. + * we will map the sound part of mac-io now so that we can probe for + * other info if necessary (early AWACS we want to read chip ids) + */ + + if (io->n_addrs < 3 || io->n_intrs < 3) { + /* OK - maybe we need to use the 'awacs' node (on earlier + * machines). + */ + if (awacs_node) { + io = awacs_node ; + if (io->n_addrs < 3 || io->n_intrs < 3) { + printk("dmasound_pmac: can't use %s" + " (%d addrs, %d intrs)\n", + io->full_name, io->n_addrs, io->n_intrs); + return -ENODEV; + } + } else { + printk("dmasound_pmac: can't use %s (%d addrs, %d intrs)\n", + io->full_name, io->n_addrs, io->n_intrs); + } + } + + if (!request_OF_resource(io, 0, NULL)) { + printk(KERN_ERR "dmasound: can't request IO resource !\n"); + return -ENODEV; + } + if (!request_OF_resource(io, 1, " (tx dma)")) { + release_OF_resource(io, 0); + printk(KERN_ERR "dmasound: can't request TX DMA resource !\n"); + return -ENODEV; + } + + if (!request_OF_resource(io, 2, " (rx dma)")) { + release_OF_resource(io, 0); + release_OF_resource(io, 1); + printk(KERN_ERR "dmasound: can't request RX DMA resource !\n"); + return -ENODEV; + } + + /* all OF versions I've seen use this value */ + if (i2s_node) + i2s = ioremap(io->addrs[0].address, 0x1000); + else + awacs = ioremap(io->addrs[0].address, 0x1000); + awacs_txdma = ioremap(io->addrs[1].address, 0x100); + awacs_rxdma = ioremap(io->addrs[2].address, 0x100); + + /* first of all make sure that the chip is powered up....*/ + pmac_call_feature(PMAC_FTR_SOUND_CHIP_ENABLE, io, 0, 1); + if (awacs_revision == AWACS_SCREAMER && awacs) + awacs_recalibrate(); + + awacs_irq = io->intrs[0].line; + awacs_tx_irq = io->intrs[1].line; + awacs_rx_irq = io->intrs[2].line; + + /* Hack for legacy crap that will be killed someday */ + awacs_node = io; + + /* if we have an awacs or screamer - probe the chip to make + * sure we have the right revision. + */ + + if (awacs_revision <= AWACS_SCREAMER){ + uint32_t temp, rev, mfg ; + /* find out the awacs revision from the chip */ + temp = in_le32(&awacs->codec_stat); + rev = (temp >> 12) & 0xf; + mfg = (temp >> 8) & 0xf; +#ifdef DEBUG_DMASOUND +printk("dmasound_pmac: Awacs/Screamer Codec Mfct: %d Rev %d\n", mfg, rev); +#endif + if (rev >= AWACS_SCREAMER) + awacs_revision = AWACS_SCREAMER ; + else + awacs_revision = rev ; + } + + dmasound.mach = machPMac; + + /* find out other bits & pieces from OF, these may be present + only on some models ... so be careful. + */ + + /* in the absence of a frame rates property we will use the defaults + */ + + if (info) { + unsigned int *prop, l; + + sound_device_id = 0; + /* device ID appears post g3 b&w */ + prop = (unsigned int *)get_property(info, "device-id", NULL); + if (prop != 0) + sound_device_id = *prop; + + /* look for a property saying what sample rates + are available */ + + prop = (unsigned int *)get_property(info, "sample-rates", &l); + if (prop == 0) + prop = (unsigned int *) get_property + (info, "output-frame-rates", &l); + + /* if it's there use it to set up frame rates */ + init_frame_rates(prop, l) ; + } + + if (awacs) + out_le32(&awacs->control, 0x11); /* set everything quiesent */ + + set_hw_byteswap(io) ; /* figure out if the h/w can do it */ + +#ifdef CONFIG_NVRAM + /* get default volume from nvram */ + vol = ((pmac_xpram_read( 8 ) & 7 ) << 1 ); +#else + vol = 0; +#endif + + /* set up tracking values */ + spk_vol = vol * 100 ; + spk_vol /= 7 ; /* get set value to a percentage */ + spk_vol |= (spk_vol << 8) ; /* equal left & right */ + line_vol = passthru_vol = spk_vol ; + + /* fill regs that are shared between AWACS & Burgundy */ + + awacs_reg[2] = vol + (vol << 6); + awacs_reg[4] = vol + (vol << 6); + awacs_reg[5] = vol + (vol << 6); /* screamer has loopthru vol control */ + awacs_reg[6] = 0; /* maybe should be vol << 3 for PCMCIA speaker */ + awacs_reg[7] = 0; + + awacs_reg[0] = MASK_MUX_CD; + awacs_reg[1] = MASK_LOOPTHRU; + + /* FIXME: Only machines with external SRS module need MASK_PAROUT */ + if (has_perch || sound_device_id == 0x5 + || /*sound_device_id == 0x8 ||*/ sound_device_id == 0xb) + awacs_reg[1] |= MASK_PAROUT0 | MASK_PAROUT1; + + switch (awacs_revision) { + case AWACS_TUMBLER: + tas_register_driver(&tas3001c_hooks); + tas_init(I2C_DRIVERID_TAS3001C, I2C_DRIVERNAME_TAS3001C); + tas_dmasound_init(); + tas_post_init(); + break ; + case AWACS_SNAPPER: + tas_register_driver(&tas3004_hooks); + tas_init(I2C_DRIVERID_TAS3004,I2C_DRIVERNAME_TAS3004); + tas_dmasound_init(); + tas_post_init(); + break; + case AWACS_DACA: + daca_init(); + break; + case AWACS_BURGUNDY: + awacs_burgundy_init(); + break ; + case AWACS_SCREAMER: + case AWACS_AWACS: + default: + load_awacs(); + break ; + } + + /* enable/set-up external modules - when we know how */ + + if (has_perch) + awacs_enable_amp(100 * 0x101); + + /* Reset dbdma channels */ + out_le32(&awacs_txdma->control, (RUN|PAUSE|FLUSH|WAKE|DEAD) << 16); + while (in_le32(&awacs_txdma->status) & RUN) + udelay(1); + out_le32(&awacs_rxdma->control, (RUN|PAUSE|FLUSH|WAKE|DEAD) << 16); + while (in_le32(&awacs_rxdma->status) & RUN) + udelay(1); + + /* Initialize beep stuff */ + if ((res=setup_beep())) + return res ; + +#ifdef CONFIG_PMAC_PBOOK + pmu_register_sleep_notifier(&awacs_sleep_notifier); +#endif /* CONFIG_PMAC_PBOOK */ + + /* Powerbooks have odd ways of enabling inputs such as + an expansion-bay CD or sound from an internal modem + or a PC-card modem. */ + if (is_pbook_3X00) { + /* + * Enable CD and PC-card sound inputs. + * This is done by reading from address + * f301a000, + 0x10 to enable the expansion-bay + * CD sound input, + 0x80 to enable the PC-card + * sound input. The 0x100 enables the SCSI bus + * terminator power. + */ + latch_base = ioremap (0xf301a000, 0x1000); + in_8(latch_base + 0x190); + + } else if (is_pbook_g3) { + struct device_node* mio; + macio_base = NULL; + for (mio = io->parent; mio; mio = mio->parent) { + if (strcmp(mio->name, "mac-io") == 0 + && mio->n_addrs > 0) { + macio_base = ioremap(mio->addrs[0].address, 0x40); + break; + } + } + /* + * Enable CD sound input. + * The relevant bits for writing to this byte are 0x8f. + * I haven't found out what the 0x80 bit does. + * For the 0xf bits, writing 3 or 7 enables the CD + * input, any other value disables it. Values + * 1, 3, 5, 7 enable the microphone. Values 0, 2, + * 4, 6, 8 - f enable the input from the modem. + * -- paulus. + */ + if (macio_base) + out_8(macio_base + 0x37, 3); + } + + if (hw_can_byteswap) + dmasound.mach.hardware_afmts = (AFMT_S16_BE | AFMT_S16_LE) ; + else + dmasound.mach.hardware_afmts = AFMT_S16_BE ; + + /* shut out chips that do output only. + * may need to extend this to machines which have no inputs - even tho' + * they use screamer - IIRC one of the powerbooks is like this. + */ + + if (awacs_revision != AWACS_DACA) { + dmasound.mach.capabilities = DSP_CAP_DUPLEX ; + dmasound.mach.record = PMacRecord ; + } + + dmasound.mach.default_hard = def_hard ; + dmasound.mach.default_soft = def_soft ; + + switch (awacs_revision) { + case AWACS_BURGUNDY: + sprintf(awacs_name, "PowerMac Burgundy ") ; + break ; + case AWACS_DACA: + sprintf(awacs_name, "PowerMac DACA ") ; + break ; + case AWACS_TUMBLER: + sprintf(awacs_name, "PowerMac Tumbler ") ; + break ; + case AWACS_SNAPPER: + sprintf(awacs_name, "PowerMac Snapper ") ; + break ; + case AWACS_SCREAMER: + sprintf(awacs_name, "PowerMac Screamer ") ; + break ; + case AWACS_AWACS: + default: + sprintf(awacs_name, "PowerMac AWACS rev %d ", awacs_revision) ; + break ; + } + + /* + * XXX: we should handle errors here, but that would mean + * rewriting the whole init code. later.. + */ + input_register_device(&awacs_beep_dev); + + return dmasound_init(); +} + +static void __exit dmasound_awacs_cleanup(void) +{ + input_unregister_device(&awacs_beep_dev); + + switch (awacs_revision) { + case AWACS_TUMBLER: + case AWACS_SNAPPER: + tas_dmasound_cleanup(); + tas_cleanup(); + break ; + case AWACS_DACA: + daca_cleanup(); + break; + } + dmasound_deinit(); + +} + +MODULE_DESCRIPTION("PowerMac built-in audio driver."); +MODULE_LICENSE("GPL"); + +module_init(dmasound_awacs_init); +module_exit(dmasound_awacs_cleanup); diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c new file mode 100644 index 0000000..c9302a1 --- /dev/null +++ b/sound/oss/dmasound/dmasound_core.c @@ -0,0 +1,1829 @@ +/* + * linux/sound/oss/dmasound/dmasound_core.c + * + * + * OSS/Free compatible Atari TT/Falcon and Amiga DMA sound driver for + * Linux/m68k + * Extended to support Power Macintosh for Linux/ppc by Paul Mackerras + * + * (c) 1995 by Michael Schlueter & Michael Marte + * + * Michael Schlueter (michael@duck.syd.de) did the basic structure of the VFS + * interface and the u-law to signed byte conversion. + * + * Michael Marte (marte@informatik.uni-muenchen.de) did the sound queue, + * /dev/mixer, /dev/sndstat and complemented the VFS interface. He would like + * to thank: + * - Michael Schlueter for initial ideas and documentation on the MFP and + * the DMA sound hardware. + * - Therapy? for their CD 'Troublegum' which really made me rock. + * + * /dev/sndstat is based on code by Hannu Savolainen, the author of the + * VoxWare family of drivers. + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive + * for more details. + * + * History: + * + * 1995/8/25 First release + * + * 1995/9/02 Roman Hodek: + * - Fixed atari_stram_alloc() call, the timer + * programming and several race conditions + * 1995/9/14 Roman Hodek: + * - After some discussion with Michael Schlueter, + * revised the interrupt disabling + * - Slightly speeded up U8->S8 translation by using + * long operations where possible + * - Added 4:3 interpolation for /dev/audio + * + * 1995/9/20 Torsten Scherer: + * - Fixed a bug in sq_write and changed /dev/audio + * converting to play at 12517Hz instead of 6258Hz. + * + * 1995/9/23 Torsten Scherer: + * - Changed sq_interrupt() and sq_play() to pre-program + * the DMA for another frame while there's still one + * running. This allows the IRQ response to be + * arbitrarily delayed and playing will still continue. + * + * 1995/10/14 Guenther Kelleter, Torsten Scherer: + * - Better support for Falcon audio (the Falcon doesn't + * raise an IRQ at the end of a frame, but at the + * beginning instead!). uses 'if (codec_dma)' in lots + * of places to simply switch between Falcon and TT + * code. + * + * 1995/11/06 Torsten Scherer: + * - Started introducing a hardware abstraction scheme + * (may perhaps also serve for Amigas?) + * - Can now play samples at almost all frequencies by + * means of a more generalized expand routine + * - Takes a good deal of care to cut data only at + * sample sizes + * - Buffer size is now a kernel runtime option + * - Implemented fsync() & several minor improvements + * Guenther Kelleter: + * - Useful hints and bug fixes + * - Cross-checked it for Falcons + * + * 1996/3/9 Geert Uytterhoeven: + * - Support added for Amiga, A-law, 16-bit little + * endian. + * - Unification to drivers/sound/dmasound.c. + * + * 1996/4/6 Martin Mitchell: + * - Updated to 1.3 kernel. + * + * 1996/6/13 Topi Kanerva: + * - Fixed things that were broken (mainly the amiga + * 14-bit routines) + * - /dev/sndstat shows now the real hardware frequency + * - The lowpass filter is disabled by default now + * + * 1996/9/25 Geert Uytterhoeven: + * - Modularization + * + * 1998/6/10 Andreas Schwab: + * - Converted to use sound_core + * + * 1999/12/28 Richard Zidlicky: + * - Added support for Q40 + * + * 2000/2/27 Geert Uytterhoeven: + * - Clean up and split the code into 4 parts: + * o dmasound_core: machine-independent code + * o dmasound_atari: Atari TT and Falcon support + * o dmasound_awacs: Apple PowerMac support + * o dmasound_paula: Amiga support + * + * 2000/3/25 Geert Uytterhoeven: + * - Integration of dmasound_q40 + * - Small clean ups + * + * 2001/01/26 [1.0] Iain Sandoe + * - make /dev/sndstat show revision & edition info. + * - since dmasound.mach.sq_setup() can fail on pmac + * its type has been changed to int and the returns + * are checked. + * [1.1] - stop missing translations from being called. + * 2001/02/08 [1.2] - remove unused translation tables & move machine- + * specific tables to low-level. + * - return correct info. for SNDCTL_DSP_GETFMTS. + * [1.3] - implement SNDCTL_DSP_GETCAPS fully. + * [1.4] - make /dev/sndstat text length usage deterministic. + * - make /dev/sndstat call to low-level + * dmasound.mach.state_info() pass max space to ll driver. + * - tidy startup banners and output info. + * [1.5] - tidy up a little (removed some unused #defines in + * dmasound.h) + * - fix up HAS_RECORD conditionalisation. + * - add record code in places it is missing... + * - change buf-sizes to bytes to allow < 1kb for pmac + * if user param entry is < 256 the value is taken to + * be in kb > 256 is taken to be in bytes. + * - make default buff/frag params conditional on + * machine to allow smaller values for pmac. + * - made the ioctls, read & write comply with the OSS + * rules on setting params. + * - added parsing of _setup() params for record. + * 2001/04/04 [1.6] - fix bug where sample rates higher than maximum were + * being reported as OK. + * - fix open() to return -EBUSY as per OSS doc. when + * audio is in use - this is independent of O_NOBLOCK. + * - fix bug where SNDCTL_DSP_POST was blocking. + */ + + /* Record capability notes 30/01/2001: + * At present these observations apply only to pmac LL driver (the only one + * that can do record, at present). However, if other LL drivers for machines + * with record are added they may apply. + * + * The fragment parameters for the record and play channels are separate. + * However, if the driver is opened O_RDWR there is no way (in the current OSS + * API) to specify their values independently for the record and playback + * channels. Since the only common factor between the input & output is the + * sample rate (on pmac) it should be possible to open /dev/dspX O_WRONLY and + * /dev/dspY O_RDONLY. The input & output channels could then have different + * characteristics (other than the first that sets sample rate claiming the + * right to set it for ever). As it stands, the format, channels, number of + * bits & sample rate are assumed to be common. In the future perhaps these + * should be the responsibility of the LL driver - and then if a card really + * does not share items between record & playback they can be specified + * separately. +*/ + +/* Thread-safeness of shared_resources notes: 31/01/2001 + * If the user opens O_RDWR and then splits record & play between two threads + * both of which inherit the fd - and then starts changing things from both + * - we will have difficulty telling. + * + * It's bad application coding - but ... + * TODO: think about how to sort this out... without bogging everything down in + * semaphores. + * + * Similarly, the OSS spec says "all changes to parameters must be between + * open() and the first read() or write(). - and a bit later on (by + * implication) "between SNDCTL_DSP_RESET and the first read() or write() after + * it". If the app is multi-threaded and this rule is broken between threads + * we will have trouble spotting it - and the fault will be rather obscure :-( + * + * We will try and put out at least a kmsg if we see it happen... but I think + * it will be quite hard to trap it with an -EXXX return... because we can't + * see the fault until after the damage is done. +*/ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/sound.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <linux/poll.h> +#include <linux/smp_lock.h> + +#include <asm/uaccess.h> + +#include "dmasound.h" + +#define DMASOUND_CORE_REVISION 1 +#define DMASOUND_CORE_EDITION 6 + + /* + * Declarations + */ + +int dmasound_catchRadius = 0; +MODULE_PARM(dmasound_catchRadius, "i"); + +static unsigned int numWriteBufs = DEFAULT_N_BUFFERS; +MODULE_PARM(numWriteBufs, "i"); +static unsigned int writeBufSize = DEFAULT_BUFF_SIZE ; /* in bytes */ +MODULE_PARM(writeBufSize, "i"); + +#ifdef HAS_RECORD +static unsigned int numReadBufs = DEFAULT_N_BUFFERS; +MODULE_PARM(numReadBufs, "i"); +static unsigned int readBufSize = DEFAULT_BUFF_SIZE; /* in bytes */ +MODULE_PARM(readBufSize, "i"); +#endif + +MODULE_LICENSE("GPL"); + +#ifdef MODULE +static int sq_unit = -1; +static int mixer_unit = -1; +static int state_unit = -1; +static int irq_installed; +#endif /* MODULE */ + +/* software implemented recording volume! */ +uint software_input_volume = SW_INPUT_VOLUME_SCALE * SW_INPUT_VOLUME_DEFAULT; +EXPORT_SYMBOL(software_input_volume); + +/* control over who can modify resources shared between play/record */ +static mode_t shared_resource_owner; +static int shared_resources_initialised; + + /* + * Mid level stuff + */ + +struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED }; + +static inline void sound_silence(void) +{ + dmasound.mach.silence(); /* _MUST_ stop DMA */ +} + +static inline int sound_set_format(int format) +{ + return dmasound.mach.setFormat(format); +} + + +static int sound_set_speed(int speed) +{ + if (speed < 0) + return dmasound.soft.speed; + + /* trap out-of-range speed settings. + at present we allow (arbitrarily) low rates - using soft + up-conversion - but we can't allow > max because there is + no soft down-conversion. + */ + if (dmasound.mach.max_dsp_speed && + (speed > dmasound.mach.max_dsp_speed)) + speed = dmasound.mach.max_dsp_speed ; + + dmasound.soft.speed = speed; + + if (dmasound.minDev == SND_DEV_DSP) + dmasound.dsp.speed = dmasound.soft.speed; + + return dmasound.soft.speed; +} + +static int sound_set_stereo(int stereo) +{ + if (stereo < 0) + return dmasound.soft.stereo; + + stereo = !!stereo; /* should be 0 or 1 now */ + + dmasound.soft.stereo = stereo; + if (dmasound.minDev == SND_DEV_DSP) + dmasound.dsp.stereo = stereo; + + return stereo; +} + +static ssize_t sound_copy_translate(TRANS *trans, const u_char __user *userPtr, + size_t userCount, u_char frame[], + ssize_t *frameUsed, ssize_t frameLeft) +{ + ssize_t (*ct_func)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + + switch (dmasound.soft.format) { + case AFMT_MU_LAW: + ct_func = trans->ct_ulaw; + break; + case AFMT_A_LAW: + ct_func = trans->ct_alaw; + break; + case AFMT_S8: + ct_func = trans->ct_s8; + break; + case AFMT_U8: + ct_func = trans->ct_u8; + break; + case AFMT_S16_BE: + ct_func = trans->ct_s16be; + break; + case AFMT_U16_BE: + ct_func = trans->ct_u16be; + break; + case AFMT_S16_LE: + ct_func = trans->ct_s16le; + break; + case AFMT_U16_LE: + ct_func = trans->ct_u16le; + break; + default: + return 0; + } + /* if the user has requested a non-existent translation don't try + to call it but just return 0 bytes moved + */ + if (ct_func) + return ct_func(userPtr, userCount, frame, frameUsed, frameLeft); + return 0; +} + + /* + * /dev/mixer abstraction + */ + +static struct { + int busy; + int modify_counter; +} mixer; + +static int mixer_open(struct inode *inode, struct file *file) +{ + if (!try_module_get(dmasound.mach.owner)) + return -ENODEV; + mixer.busy = 1; + return 0; +} + +static int mixer_release(struct inode *inode, struct file *file) +{ + lock_kernel(); + mixer.busy = 0; + module_put(dmasound.mach.owner); + unlock_kernel(); + return 0; +} +static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, + u_long arg) +{ + if (_SIOC_DIR(cmd) & _SIOC_WRITE) + mixer.modify_counter++; + switch (cmd) { + case OSS_GETVERSION: + return IOCTL_OUT(arg, SOUND_VERSION); + case SOUND_MIXER_INFO: + { + mixer_info info; + memset(&info, 0, sizeof(info)); + strlcpy(info.id, dmasound.mach.name2, sizeof(info.id)); + strlcpy(info.name, dmasound.mach.name2, sizeof(info.name)); + info.modify_counter = mixer.modify_counter; + if (copy_to_user((void __user *)arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } + } + if (dmasound.mach.mixer_ioctl) + return dmasound.mach.mixer_ioctl(cmd, arg); + return -EINVAL; +} + +static struct file_operations mixer_fops = +{ + .owner = THIS_MODULE, + .llseek = no_llseek, + .ioctl = mixer_ioctl, + .open = mixer_open, + .release = mixer_release, +}; + +static void mixer_init(void) +{ +#ifndef MODULE + int mixer_unit; +#endif + mixer_unit = register_sound_mixer(&mixer_fops, -1); + if (mixer_unit < 0) + return; + + mixer.busy = 0; + dmasound.treble = 0; + dmasound.bass = 0; + if (dmasound.mach.mixer_init) + dmasound.mach.mixer_init(); +} + + + /* + * Sound queue stuff, the heart of the driver + */ + +struct sound_queue dmasound_write_sq; +static void sq_reset_output(void) ; +#ifdef HAS_RECORD +struct sound_queue dmasound_read_sq; +static void sq_reset_input(void) ; +#endif + +static int sq_allocate_buffers(struct sound_queue *sq, int num, int size) +{ + int i; + + if (sq->buffers) + return 0; + sq->numBufs = num; + sq->bufSize = size; + sq->buffers = kmalloc (num * sizeof(char *), GFP_KERNEL); + if (!sq->buffers) + return -ENOMEM; + for (i = 0; i < num; i++) { + sq->buffers[i] = dmasound.mach.dma_alloc(size, GFP_KERNEL); + if (!sq->buffers[i]) { + while (i--) + dmasound.mach.dma_free(sq->buffers[i], size); + kfree(sq->buffers); + sq->buffers = NULL; + return -ENOMEM; + } + } + return 0; +} + +static void sq_release_buffers(struct sound_queue *sq) +{ + int i; + + if (sq->buffers) { + for (i = 0; i < sq->numBufs; i++) + dmasound.mach.dma_free(sq->buffers[i], sq->bufSize); + kfree(sq->buffers); + sq->buffers = NULL; + } +} + + +static int sq_setup(struct sound_queue *sq) +{ + int (*setup_func)(void) = NULL; + int hard_frame ; + + if (sq->locked) { /* are we already set? - and not changeable */ +#ifdef DEBUG_DMASOUND +printk("dmasound_core: tried to sq_setup a locked queue\n") ; +#endif + return -EINVAL ; + } + sq->locked = 1 ; /* don't think we have a race prob. here _check_ */ + + /* make sure that the parameters are set up + This should have been done already... + */ + + dmasound.mach.init(); + + /* OK. If the user has set fragment parameters explicitly, then we + should leave them alone... as long as they are valid. + Invalid user fragment params can occur if we allow the whole buffer + to be used when the user requests the fragments sizes (with no soft + x-lation) and then the user subsequently sets a soft x-lation that + requires increased internal buffering. + + Othwerwise (if the user did not set them) OSS says that we should + select frag params on the basis of 0.5 s output & 0.1 s input + latency. (TODO. For now we will copy in the defaults.) + */ + + if (sq->user_frags <= 0) { + sq->max_count = sq->numBufs ; + sq->max_active = sq->numBufs ; + sq->block_size = sq->bufSize; + /* set up the user info */ + sq->user_frags = sq->numBufs ; + sq->user_frag_size = sq->bufSize ; + sq->user_frag_size *= + (dmasound.soft.size * (dmasound.soft.stereo+1) ) ; + sq->user_frag_size /= + (dmasound.hard.size * (dmasound.hard.stereo+1) ) ; + } else { + /* work out requested block size */ + sq->block_size = sq->user_frag_size ; + sq->block_size *= + (dmasound.hard.size * (dmasound.hard.stereo+1) ) ; + sq->block_size /= + (dmasound.soft.size * (dmasound.soft.stereo+1) ) ; + /* the user wants to write frag-size chunks */ + sq->block_size *= dmasound.hard.speed ; + sq->block_size /= dmasound.soft.speed ; + /* this only works for size values which are powers of 2 */ + hard_frame = + (dmasound.hard.size * (dmasound.hard.stereo+1))/8 ; + sq->block_size += (hard_frame - 1) ; + sq->block_size &= ~(hard_frame - 1) ; /* make sure we are aligned */ + /* let's just check for obvious mistakes */ + if ( sq->block_size <= 0 || sq->block_size > sq->bufSize) { +#ifdef DEBUG_DMASOUND +printk("dmasound_core: invalid frag size (user set %d)\n", sq->user_frag_size) ; +#endif + sq->block_size = sq->bufSize ; + } + if ( sq->user_frags <= sq->numBufs ) { + sq->max_count = sq->user_frags ; + /* if user has set max_active - then use it */ + sq->max_active = (sq->max_active <= sq->max_count) ? + sq->max_active : sq->max_count ; + } else { +#ifdef DEBUG_DMASOUND +printk("dmasound_core: invalid frag count (user set %d)\n", sq->user_frags) ; +#endif + sq->max_count = + sq->max_active = sq->numBufs ; + } + } + sq->front = sq->count = sq->rear_size = 0; + sq->syncing = 0; + sq->active = 0; + + if (sq == &write_sq) { + sq->rear = -1; + setup_func = dmasound.mach.write_sq_setup; + } +#ifdef HAS_RECORD + else { + sq->rear = 0; + setup_func = dmasound.mach.read_sq_setup; + } +#endif + if (setup_func) + return setup_func(); + return 0 ; +} + +static inline void sq_play(void) +{ + dmasound.mach.play(); +} + +static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft, + loff_t *ppos) +{ + ssize_t uWritten = 0; + u_char *dest; + ssize_t uUsed = 0, bUsed, bLeft; + unsigned long flags ; + + /* ++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; + + /* implement any changes we have made to the soft/hard params. + this is not satisfactory really, all we have done up to now is to + say what we would like - there hasn't been any real checking of capability + */ + + if (shared_resources_initialised == 0) { + dmasound.mach.init() ; + shared_resources_initialised = 1 ; + } + + /* set up the sq if it is not already done. This may seem a dumb place + to do it - but it is what OSS requires. It means that write() can + return memory allocation errors. To avoid this possibility use the + GETBLKSIZE or GETOSPACE ioctls (after you've fiddled with all the + params you want to change) - these ioctls also force the setup. + */ + + if (write_sq.locked == 0) { + if ((uWritten = sq_setup(&write_sq)) < 0) return uWritten ; + uWritten = 0 ; + } + +/* FIXME: I think that this may be the wrong behaviour when we get strapped + for time and the cpu is close to being (or actually) behind in sending data. + - because we've lost the time that the N samples, already in the buffer, + would have given us to get here with the next lot from the user. +*/ + /* The interrupt doesn't start to play the last, incomplete frame. + * Thus we can append to it without disabling the interrupts! (Note + * also that write_sq.rear isn't affected by the interrupt.) + */ + + /* as of 1.6 this behaviour changes if SNDCTL_DSP_POST has been issued: + this will mimic the behaviour of syncing and allow the sq_play() to + queue a partial fragment. Since sq_play() may/will be called from + the IRQ handler - at least on Pmac we have to deal with it. + The strategy - possibly not optimum - is to kill _POST status if we + get here. This seems, at least, reasonable - in the sense that POST + is supposed to indicate that we might not write before the queue + is drained - and if we get here in time then it does not apply. + */ + + spin_lock_irqsave(&dmasound.lock, flags); + write_sq.syncing &= ~2 ; /* take out POST status */ + spin_unlock_irqrestore(&dmasound.lock, flags); + + if (write_sq.count > 0 && + (bLeft = write_sq.block_size-write_sq.rear_size) > 0) { + dest = write_sq.buffers[write_sq.rear]; + bUsed = write_sq.rear_size; + uUsed = sound_copy_translate(dmasound.trans_write, src, uLeft, + dest, &bUsed, bLeft); + if (uUsed <= 0) + return uUsed; + src += uUsed; + uWritten += uUsed; + uLeft = (uUsed <= uLeft) ? (uLeft - uUsed) : 0 ; /* paranoia */ + write_sq.rear_size = bUsed; + } + + while (uLeft) { + while (write_sq.count >= write_sq.max_active) { + sq_play(); + if (write_sq.open_mode & O_NONBLOCK) + return uWritten > 0 ? uWritten : -EAGAIN; + SLEEP(write_sq.action_queue); + if (signal_pending(current)) + return uWritten > 0 ? uWritten : -EINTR; + } + + /* Here, we can avoid disabling the interrupt by first + * copying and translating the data, and then updating + * the write_sq variables. Until this is done, the interrupt + * won't see the new frame and we can work on it + * undisturbed. + */ + + dest = write_sq.buffers[(write_sq.rear+1) % write_sq.max_count]; + bUsed = 0; + bLeft = write_sq.block_size; + uUsed = sound_copy_translate(dmasound.trans_write, src, uLeft, + dest, &bUsed, bLeft); + if (uUsed <= 0) + break; + src += uUsed; + uWritten += uUsed; + uLeft = (uUsed <= uLeft) ? (uLeft - uUsed) : 0 ; /* paranoia */ + if (bUsed) { + write_sq.rear = (write_sq.rear+1) % write_sq.max_count; + write_sq.rear_size = bUsed; + write_sq.count++; + } + } /* uUsed may have been 0 */ + + sq_play(); + + return uUsed < 0? uUsed: uWritten; +} + +static unsigned int sq_poll(struct file *file, struct poll_table_struct *wait) +{ + unsigned int mask = 0; + int retVal; + + if (write_sq.locked == 0) { + if ((retVal = sq_setup(&write_sq)) < 0) + return retVal; + return 0; + } + if (file->f_mode & FMODE_WRITE ) + poll_wait(file, &write_sq.action_queue, wait); +#ifdef HAS_RECORD + if (file->f_mode & FMODE_READ) + poll_wait(file, &read_sq.action_queue, wait); + if (file->f_mode & FMODE_READ) + if (read_sq.block_size - read_sq.rear_size > 0) + mask |= POLLIN | POLLRDNORM; +#endif + if (file->f_mode & FMODE_WRITE) + if (write_sq.count < write_sq.max_active || write_sq.block_size - write_sq.rear_size > 0) + mask |= POLLOUT | POLLWRNORM; + return mask; + +} + +#ifdef HAS_RECORD + /* + * 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 buffer we are currently returning to the user. + + * This level (> [1.5]) doesn't care what strategy the LL driver uses with + * DMA on over-run. It can leave it running (and keep active == 1) or it + * can kill it and set active == 0 in which case this routine will spot + * it and restart the DMA. + */ + +static ssize_t sq_read(struct file *file, char __user *dst, size_t uLeft, + loff_t *ppos) +{ + + ssize_t uRead, bLeft, bUsed, uUsed; + + if (uLeft == 0) + return 0; + + /* cater for the compatibility mode - record compiled in but no LL */ + if (dmasound.mach.record == NULL) + return -EINVAL ; + + /* see comment in sq_write() + */ + + if( shared_resources_initialised == 0) { + dmasound.mach.init() ; + shared_resources_initialised = 1 ; + } + + /* set up the sq if it is not already done. see comments in sq_write(). + */ + + if (read_sq.locked == 0) { + if ((uRead = sq_setup(&read_sq)) < 0) + return uRead ; + } + + uRead = 0; + + /* Move what the user requests, depending upon other options. + */ + while (uLeft > 0) { + + /* we happened to get behind and the LL driver killed DMA + then we should set it going again. This also sets it + going the first time through. + */ + if ( !read_sq.active ) + dmasound.mach.record(); + + /* When front == rear, the DMA is not done yet. + */ + while (read_sq.front == read_sq.rear) { + if (read_sq.open_mode & O_NONBLOCK) { + return uRead > 0 ? uRead : -EAGAIN; + } + SLEEP(read_sq.action_queue); + if (signal_pending(current)) + 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(dmasound.trans_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; +} +#endif /* HAS_RECORD */ + +static inline void sq_init_waitqueue(struct sound_queue *sq) +{ + init_waitqueue_head(&sq->action_queue); + init_waitqueue_head(&sq->open_queue); + init_waitqueue_head(&sq->sync_queue); + sq->busy = 0; +} + +#if 0 /* blocking open() */ +static inline void sq_wake_up(struct sound_queue *sq, struct file *file, + mode_t mode) +{ + if (file->f_mode & mode) { + sq->busy = 0; /* CHECK: IS THIS OK??? */ + WAKE_UP(sq->open_queue); + } +} +#endif + +static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, + int numbufs, int bufsize) +{ + int rc = 0; + + if (file->f_mode & mode) { + if (sq->busy) { +#if 0 /* blocking open() */ + rc = -EBUSY; + if (file->f_flags & O_NONBLOCK) + return rc; + rc = -EINTR; + while (sq->busy) { + SLEEP(sq->open_queue); + if (signal_pending(current)) + return rc; + } + rc = 0; +#else + /* OSS manual says we will return EBUSY regardless + of O_NOBLOCK. + */ + return -EBUSY ; +#endif + } + sq->busy = 1; /* Let's play spot-the-race-condition */ + + /* allocate the default number & size of buffers. + (i.e. specified in _setup() or as module params) + can't be changed at the moment - but _could_ be perhaps + in the setfragments ioctl. + */ + if (( rc = sq_allocate_buffers(sq, numbufs, bufsize))) { +#if 0 /* blocking open() */ + sq_wake_up(sq, file, mode); +#else + sq->busy = 0 ; +#endif + return rc; + } + + sq->open_mode = file->f_mode; + } + return rc; +} + +#define write_sq_init_waitqueue() sq_init_waitqueue(&write_sq) +#if 0 /* blocking open() */ +#define write_sq_wake_up(file) sq_wake_up(&write_sq, file, FMODE_WRITE) +#endif +#define write_sq_release_buffers() sq_release_buffers(&write_sq) +#define write_sq_open(file) \ + sq_open2(&write_sq, file, FMODE_WRITE, numWriteBufs, writeBufSize ) + +#ifdef HAS_RECORD +#define read_sq_init_waitqueue() sq_init_waitqueue(&read_sq) +#if 0 /* blocking open() */ +#define read_sq_wake_up(file) sq_wake_up(&read_sq, file, FMODE_READ) +#endif +#define read_sq_release_buffers() sq_release_buffers(&read_sq) +#define read_sq_open(file) \ + sq_open2(&read_sq, file, FMODE_READ, numReadBufs, readBufSize ) +#else +#define read_sq_init_waitqueue() do {} while (0) +#if 0 /* blocking open() */ +#define read_sq_wake_up(file) do {} while (0) +#endif +#define read_sq_release_buffers() do {} while (0) +#define sq_reset_input() do {} while (0) +#endif + +static int sq_open(struct inode *inode, struct file *file) +{ + int rc; + + if (!try_module_get(dmasound.mach.owner)) + return -ENODEV; + + rc = write_sq_open(file); /* checks the f_mode */ + if (rc) + goto out; +#ifdef HAS_RECORD + if (dmasound.mach.record) { + rc = read_sq_open(file); /* checks the f_mode */ + if (rc) + goto out; + } else { /* no record function installed; in compat mode */ + if (file->f_mode & FMODE_READ) { + /* TODO: if O_RDWR, release any resources grabbed by write part */ + rc = -ENXIO; + goto out; + } + } +#else /* !HAS_RECORD */ + if (file->f_mode & FMODE_READ) { + /* TODO: if O_RDWR, release any resources grabbed by write part */ + rc = -ENXIO ; /* I think this is what is required by open(2) */ + goto out; + } +#endif /* HAS_RECORD */ + + if (dmasound.mach.sq_open) + dmasound.mach.sq_open(file->f_mode); + + /* CHECK whether this is sensible - in the case that dsp0 could be opened + O_RDONLY and dsp1 could be opened O_WRONLY + */ + + dmasound.minDev = iminor(inode) & 0x0f; + + /* OK. - we should make some attempt at consistency. At least the H'ware + options should be set with a valid mode. We will make it that the LL + driver must supply defaults for hard & soft params. + */ + + if (shared_resource_owner == 0) { + /* you can make this AFMT_U8/mono/8K if you want to mimic old + OSS behaviour - while we still have soft translations ;-) */ + dmasound.soft = dmasound.mach.default_soft ; + dmasound.dsp = dmasound.mach.default_soft ; + dmasound.hard = dmasound.mach.default_hard ; + } + +#ifndef DMASOUND_STRICT_OSS_COMPLIANCE + /* none of the current LL drivers can actually do this "native" at the moment + OSS does not really require us to supply /dev/audio if we can't do it. + */ + if (dmasound.minDev == SND_DEV_AUDIO) { + sound_set_speed(8000); + sound_set_stereo(0); + sound_set_format(AFMT_MU_LAW); + } +#endif + + return 0; + out: + module_put(dmasound.mach.owner); + return rc; +} + +static void sq_reset_output(void) +{ + sound_silence(); /* this _must_ stop DMA, we might be about to lose the buffers */ + write_sq.active = 0; + write_sq.count = 0; + write_sq.rear_size = 0; + /* write_sq.front = (write_sq.rear+1) % write_sq.max_count;*/ + write_sq.front = 0 ; + write_sq.rear = -1 ; /* same as for set-up */ + + /* OK - we can unlock the parameters and fragment settings */ + write_sq.locked = 0 ; + write_sq.user_frags = 0 ; + write_sq.user_frag_size = 0 ; +} + +#ifdef HAS_RECORD + +static void sq_reset_input(void) +{ + if (dmasound.mach.record && read_sq.active) { + if (dmasound.mach.abort_read) { /* this routine must really be present */ + read_sq.syncing = 1 ; + /* this can use the read_sq.sync_queue to sleep if + necessary - it should not return until DMA + is really stopped - because we might deallocate + the buffers as the next action... + */ + dmasound.mach.abort_read() ; + } else { + printk(KERN_ERR + "dmasound_core: %s has no abort_read()!! all bets are off\n", + dmasound.mach.name) ; + } + } + read_sq.syncing = + read_sq.active = + read_sq.front = + read_sq.count = + read_sq.rear = 0 ; + + /* OK - we can unlock the parameters and fragment settings */ + read_sq.locked = 0 ; + read_sq.user_frags = 0 ; + read_sq.user_frag_size = 0 ; +} + +#endif + +static void sq_reset(void) +{ + sq_reset_output() ; + sq_reset_input() ; + /* we could consider resetting the shared_resources_owner here... but I + think it is probably still rather non-obvious to application writer + */ + + /* we release everything else though */ + shared_resources_initialised = 0 ; +} + +static int sq_fsync(struct file *filp, struct dentry *dentry) +{ + int rc = 0; + int timeout = 5; + + write_sq.syncing |= 1; + sq_play(); /* there may be an incomplete frame waiting */ + + while (write_sq.active) { + SLEEP(write_sq.sync_queue); + if (signal_pending(current)) { + /* While waiting for audio output to drain, an + * interrupt occurred. Stop audio output immediately + * and clear the queue. */ + sq_reset_output(); + rc = -EINTR; + break; + } + if (!--timeout) { + printk(KERN_WARNING "dmasound: Timeout draining output\n"); + sq_reset_output(); + rc = -EIO; + break; + } + } + + /* flag no sync regardless of whether we had a DSP_POST or not */ + write_sq.syncing = 0 ; + return rc; +} + +static int sq_release(struct inode *inode, struct file *file) +{ + int rc = 0; + + lock_kernel(); + +#ifdef HAS_RECORD + /* probably best to do the read side first - so that time taken to do it + overlaps with playing any remaining output samples. + */ + if (file->f_mode & FMODE_READ) { + sq_reset_input() ; /* make sure dma is stopped and all is quiet */ + read_sq_release_buffers(); + read_sq.busy = 0; + } +#endif + + if (file->f_mode & FMODE_WRITE) { + if (write_sq.busy) + rc = sq_fsync(file, file->f_dentry); + + sq_reset_output() ; /* make sure dma is stopped and all is quiet */ + write_sq_release_buffers(); + write_sq.busy = 0; + } + + if (file->f_mode & shared_resource_owner) { /* it's us that has them */ + shared_resource_owner = 0 ; + shared_resources_initialised = 0 ; + dmasound.hard = dmasound.mach.default_hard ; + } + + module_put(dmasound.mach.owner); + +#if 0 /* blocking open() */ + /* Wake up a process waiting for the queue being released. + * Note: There may be several processes waiting for a call + * to open() returning. */ + + /* Iain: hmm I don't understand this next comment ... */ + /* There is probably a DOS atack here. They change the mode flag. */ + /* XXX add check here,*/ + read_sq_wake_up(file); /* checks f_mode */ + write_sq_wake_up(file); /* checks f_mode */ +#endif /* blocking open() */ + + unlock_kernel(); + + return rc; +} + +/* here we see if we have a right to modify format, channels, size and so on + if no-one else has claimed it already then we do... + + TODO: We might change this to mask O_RDWR such that only one or the other channel + is the owner - if we have problems. +*/ + +static int shared_resources_are_mine(mode_t md) +{ + if (shared_resource_owner) + return (shared_resource_owner & md ) ; + else { + shared_resource_owner = md ; + return 1 ; + } +} + +/* if either queue is locked we must deny the right to change shared params +*/ + +static int queues_are_quiescent(void) +{ +#ifdef HAS_RECORD + if (dmasound.mach.record) + if (read_sq.locked) + return 0 ; +#endif + if (write_sq.locked) + return 0 ; + return 1 ; +} + +/* check and set a queue's fragments per user's wishes... + we will check against the pre-defined literals and the actual sizes. + This is a bit fraught - because soft translations can mess with our + buffer requirements *after* this call - OSS says "call setfrags first" +*/ + +/* It is possible to replace all the -EINVAL returns with an override that + just puts the allowable value in. This may be what many OSS apps require +*/ + +static int set_queue_frags(struct sound_queue *sq, int bufs, int size) +{ + if (sq->locked) { +#ifdef DEBUG_DMASOUND +printk("dmasound_core: tried to set_queue_frags on a locked queue\n") ; +#endif + return -EINVAL ; + } + + if ((size < MIN_FRAG_SIZE) || (size > MAX_FRAG_SIZE)) + return -EINVAL ; + size = (1<<size) ; /* now in bytes */ + if (size > sq->bufSize) + return -EINVAL ; /* this might still not work */ + + if (bufs <= 0) + return -EINVAL ; + if (bufs > sq->numBufs) /* the user is allowed say "don't care" with 0x7fff */ + bufs = sq->numBufs ; + + /* there is, currently, no way to specify max_active separately + from max_count. This could be a LL driver issue - I guess + if there is a requirement for these values to be different then + we will have to pass that info. up to this level. + */ + sq->user_frags = + sq->max_active = bufs ; + sq->user_frag_size = size ; + + return 0 ; +} + +static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, + u_long arg) +{ + int val, result; + u_long fmt; + int data; + int size, nbufs; + audio_buf_info info; + + switch (cmd) { + case SNDCTL_DSP_RESET: + sq_reset(); + return 0; + break ; + case SNDCTL_DSP_GETFMTS: + fmt = dmasound.mach.hardware_afmts ; /* this is what OSS says.. */ + return IOCTL_OUT(arg, fmt); + break ; + case SNDCTL_DSP_GETBLKSIZE: + /* this should tell the caller about bytes that the app can + read/write - the app doesn't care about our internal buffers. + We force sq_setup() here as per OSS 1.1 (which should + compute the values necessary). + Since there is no mechanism to specify read/write separately, for + fds opened O_RDWR, the write_sq values will, arbitrarily, overwrite + the read_sq ones. + */ + size = 0 ; +#ifdef HAS_RECORD + if (dmasound.mach.record && (file->f_mode & FMODE_READ)) { + if ( !read_sq.locked ) + sq_setup(&read_sq) ; /* set params */ + size = read_sq.user_frag_size ; + } +#endif + if (file->f_mode & FMODE_WRITE) { + if ( !write_sq.locked ) + sq_setup(&write_sq) ; + size = write_sq.user_frag_size ; + } + return IOCTL_OUT(arg, size); + break ; + case SNDCTL_DSP_POST: + /* all we are going to do is to tell the LL that any + partial frags can be queued for output. + The LL will have to clear this flag when last output + is queued. + */ + write_sq.syncing |= 0x2 ; + sq_play() ; + return 0 ; + case SNDCTL_DSP_SYNC: + /* This call, effectively, has the same behaviour as SNDCTL_DSP_RESET + except that it waits for output to finish before resetting + everything - read, however, is killed imediately. + */ + result = 0 ; + if ((file->f_mode & FMODE_READ) && dmasound.mach.record) + sq_reset_input() ; + if (file->f_mode & FMODE_WRITE) { + result = sq_fsync(file, file->f_dentry); + sq_reset_output() ; + } + /* if we are the shared resource owner then release them */ + if (file->f_mode & shared_resource_owner) + shared_resources_initialised = 0 ; + return result ; + break ; + case SOUND_PCM_READ_RATE: + return IOCTL_OUT(arg, dmasound.soft.speed); + case SNDCTL_DSP_SPEED: + /* changing this on the fly will have weird effects on the sound. + Where there are rate conversions implemented in soft form - it + will cause the _ctx_xxx() functions to be substituted. + However, there doesn't appear to be any reason to dis-allow it from + a driver pov. + */ + if (shared_resources_are_mine(file->f_mode)) { + IOCTL_IN(arg, data); + data = sound_set_speed(data) ; + shared_resources_initialised = 0 ; + return IOCTL_OUT(arg, data); + } else + return -EINVAL ; + break ; + /* OSS says these next 4 actions are undefined when the device is + busy/active - we will just return -EINVAL. + To be allowed to change one - (a) you have to own the right + (b) the queue(s) must be quiescent + */ + case SNDCTL_DSP_STEREO: + if (shared_resources_are_mine(file->f_mode) && + queues_are_quiescent()) { + IOCTL_IN(arg, data); + shared_resources_initialised = 0 ; + return IOCTL_OUT(arg, sound_set_stereo(data)); + } else + return -EINVAL ; + break ; + case SOUND_PCM_WRITE_CHANNELS: + if (shared_resources_are_mine(file->f_mode) && + queues_are_quiescent()) { + IOCTL_IN(arg, data); + /* the user might ask for 20 channels, we will return 1 or 2 */ + shared_resources_initialised = 0 ; + return IOCTL_OUT(arg, sound_set_stereo(data-1)+1); + } else + return -EINVAL ; + break ; + case SNDCTL_DSP_SETFMT: + if (shared_resources_are_mine(file->f_mode) && + queues_are_quiescent()) { + int format; + IOCTL_IN(arg, data); + shared_resources_initialised = 0 ; + format = sound_set_format(data); + result = IOCTL_OUT(arg, format); + if (result < 0) + return result; + if (format != data && data != AFMT_QUERY) + return -EINVAL; + return 0; + } else + return -EINVAL ; + case SNDCTL_DSP_SUBDIVIDE: + return -EINVAL ; + case SNDCTL_DSP_SETFRAGMENT: + /* we can do this independently for the two queues - with the + proviso that for fds opened O_RDWR we cannot separate the + actions and both queues will be set per the last call. + NOTE: this does *NOT* actually set the queue up - merely + registers our intentions. + */ + IOCTL_IN(arg, data); + result = 0 ; + nbufs = (data >> 16) & 0x7fff ; /* 0x7fff is 'use maximum' */ + size = data & 0xffff; +#ifdef HAS_RECORD + if ((file->f_mode & FMODE_READ) && dmasound.mach.record) { + result = set_queue_frags(&read_sq, nbufs, size) ; + if (result) + return result ; + } +#endif + if (file->f_mode & FMODE_WRITE) { + result = set_queue_frags(&write_sq, nbufs, size) ; + if (result) + return result ; + } + /* NOTE: this return value is irrelevant - OSS specifically says that + the value is 'random' and that the user _must_ check the actual + frags values using SNDCTL_DSP_GETBLKSIZE or similar */ + return IOCTL_OUT(arg, data); + break ; + case SNDCTL_DSP_GETOSPACE: + /* + */ + if (file->f_mode & FMODE_WRITE) { + if ( !write_sq.locked ) + sq_setup(&write_sq) ; + info.fragments = write_sq.max_active - write_sq.count; + info.fragstotal = write_sq.max_active; + info.fragsize = write_sq.user_frag_size; + info.bytes = info.fragments * info.fragsize; + if (copy_to_user((void __user *)arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } else + return -EINVAL ; + break ; + case SNDCTL_DSP_GETCAPS: + val = dmasound.mach.capabilities & 0xffffff00; + return IOCTL_OUT(arg,val); + + default: + return mixer_ioctl(inode, file, cmd, arg); + } + return -EINVAL; +} + +static struct file_operations sq_fops = +{ + .owner = THIS_MODULE, + .llseek = no_llseek, + .write = sq_write, + .poll = sq_poll, + .ioctl = sq_ioctl, + .open = sq_open, + .release = sq_release, +#ifdef HAS_RECORD + .read = NULL /* default to no read for compat mode */ +#endif +}; + +static int sq_init(void) +{ +#ifndef MODULE + int sq_unit; +#endif + +#ifdef HAS_RECORD + if (dmasound.mach.record) + sq_fops.read = sq_read ; +#endif + sq_unit = register_sound_dsp(&sq_fops, -1); + if (sq_unit < 0) { + printk(KERN_ERR "dmasound_core: couldn't register fops\n") ; + return sq_unit ; + } + + write_sq_init_waitqueue(); + read_sq_init_waitqueue(); + + /* These parameters will be restored for every clean open() + * in the case of multiple open()s (e.g. dsp0 & dsp1) they + * will be set so long as the shared resources have no owner. + */ + + if (shared_resource_owner == 0) { + dmasound.soft = dmasound.mach.default_soft ; + dmasound.hard = dmasound.mach.default_hard ; + dmasound.dsp = dmasound.mach.default_soft ; + shared_resources_initialised = 0 ; + } + return 0 ; +} + + + /* + * /dev/sndstat + */ + +/* we allow more space for record-enabled because there are extra output lines. + the number here must include the amount we are prepared to give to the low-level + driver. +*/ + +#ifdef HAS_RECORD +#define STAT_BUFF_LEN 1024 +#else +#define STAT_BUFF_LEN 768 +#endif + +/* this is how much space we will allow the low-level driver to use + in the stat buffer. Currently, 2 * (80 character line + <NL>). + We do not police this (it is up to the ll driver to be honest). +*/ + +#define LOW_LEVEL_STAT_ALLOC 162 + +static struct { + int busy; + char buf[STAT_BUFF_LEN]; /* state.buf should not overflow! */ + int len, ptr; +} state; + +/* publish this function for use by low-level code, if required */ + +char *get_afmt_string(int afmt) +{ + switch(afmt) { + case AFMT_MU_LAW: + return "mu-law"; + break; + case AFMT_A_LAW: + return "A-law"; + break; + case AFMT_U8: + return "unsigned 8 bit"; + break; + case AFMT_S8: + return "signed 8 bit"; + break; + case AFMT_S16_BE: + return "signed 16 bit BE"; + break; + case AFMT_U16_BE: + return "unsigned 16 bit BE"; + break; + case AFMT_S16_LE: + return "signed 16 bit LE"; + break; + case AFMT_U16_LE: + return "unsigned 16 bit LE"; + break; + case 0: + return "format not set" ; + break ; + default: + break ; + } + return "ERROR: Unsupported AFMT_XXXX code" ; +} + +static int state_open(struct inode *inode, struct file *file) +{ + char *buffer = state.buf; + int len = 0; + + if (state.busy) + return -EBUSY; + + if (!try_module_get(dmasound.mach.owner)) + return -ENODEV; + state.ptr = 0; + state.busy = 1; + + len += sprintf(buffer+len, "%sDMA sound driver rev %03d :\n", + dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) + + ((dmasound.mach.version>>8) & 0x0f)); + len += sprintf(buffer+len, + "Core driver edition %02d.%02d : %s driver edition %02d.%02d\n", + DMASOUND_CORE_REVISION, DMASOUND_CORE_EDITION, dmasound.mach.name2, + (dmasound.mach.version >> 8), (dmasound.mach.version & 0xff)) ; + + /* call the low-level module to fill in any stat info. that it has + if present. Maximum buffer usage is specified. + */ + + if (dmasound.mach.state_info) + len += dmasound.mach.state_info(buffer+len, + (size_t) LOW_LEVEL_STAT_ALLOC) ; + + /* make usage of the state buffer as deterministic as poss. + exceptional conditions could cause overrun - and this is flagged as + a kernel error. + */ + + /* formats and settings */ + + len += sprintf(buffer+len,"\t\t === Formats & settings ===\n") ; + len += sprintf(buffer+len,"Parameter %20s%20s\n","soft","hard") ; + len += sprintf(buffer+len,"Format :%20s%20s\n", + get_afmt_string(dmasound.soft.format), + get_afmt_string(dmasound.hard.format)); + + len += sprintf(buffer+len,"Samp Rate:%14d s/sec%14d s/sec\n", + dmasound.soft.speed, dmasound.hard.speed); + + len += sprintf(buffer+len,"Channels :%20s%20s\n", + dmasound.soft.stereo ? "stereo" : "mono", + dmasound.hard.stereo ? "stereo" : "mono" ); + + /* sound queue status */ + + len += sprintf(buffer+len,"\t\t === Sound Queue status ===\n"); + len += sprintf(buffer+len,"Allocated:%8s%6s\n","Buffers","Size") ; + len += sprintf(buffer+len,"%9s:%8d%6d\n", + "write", write_sq.numBufs, write_sq.bufSize) ; +#ifdef HAS_RECORD + if (dmasound.mach.record) + len += sprintf(buffer+len,"%9s:%8d%6d\n", + "read", read_sq.numBufs, read_sq.bufSize) ; +#endif + len += sprintf(buffer+len, + "Current : MaxFrg FragSiz MaxAct Frnt Rear " + "Cnt RrSize A B S L xruns\n") ; + len += sprintf(buffer+len,"%9s:%7d%8d%7d%5d%5d%4d%7d%2d%2d%2d%2d%7d\n", + "write", write_sq.max_count, write_sq.block_size, + write_sq.max_active, write_sq.front, write_sq.rear, + write_sq.count, write_sq.rear_size, write_sq.active, + write_sq.busy, write_sq.syncing, write_sq.locked, write_sq.xruns) ; +#ifdef HAS_RECORD + if (dmasound.mach.record) + len += sprintf(buffer+len,"%9s:%7d%8d%7d%5d%5d%4d%7d%2d%2d%2d%2d%7d\n", + "read", read_sq.max_count, read_sq.block_size, + read_sq.max_active, read_sq.front, read_sq.rear, + read_sq.count, read_sq.rear_size, read_sq.active, + read_sq.busy, read_sq.syncing, read_sq.locked, read_sq.xruns) ; +#endif +#ifdef DEBUG_DMASOUND +printk("dmasound: stat buffer used %d bytes\n", len) ; +#endif + + if (len >= STAT_BUFF_LEN) + printk(KERN_ERR "dmasound_core: stat buffer overflowed!\n"); + + state.len = len; + return 0; +} + +static int state_release(struct inode *inode, struct file *file) +{ + lock_kernel(); + state.busy = 0; + module_put(dmasound.mach.owner); + unlock_kernel(); + return 0; +} + +static ssize_t state_read(struct file *file, char __user *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 struct file_operations state_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .read = state_read, + .open = state_open, + .release = state_release, +}; + +static int state_init(void) +{ +#ifndef MODULE + int state_unit; +#endif + state_unit = register_sound_special(&state_fops, SND_DEV_STATUS); + if (state_unit < 0) + return state_unit ; + state.busy = 0; + return 0 ; +} + + + /* + * Config & Setup + * + * This function is called by _one_ chipset-specific driver + */ + +int dmasound_init(void) +{ + int res ; +#ifdef MODULE + if (irq_installed) + return -EBUSY; +#endif + + /* Set up sound queue, /dev/audio and /dev/dsp. */ + + /* Set default settings. */ + if ((res = sq_init()) < 0) + return res ; + + /* Set up /dev/sndstat. */ + if ((res = state_init()) < 0) + return res ; + + /* Set up /dev/mixer. */ + mixer_init(); + + if (!dmasound.mach.irqinit()) { + printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n"); + return -ENODEV; + } +#ifdef MODULE + irq_installed = 1; +#endif + + printk(KERN_INFO "%s DMA sound driver rev %03d installed\n", + dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) + + ((dmasound.mach.version>>8) & 0x0f)); + printk(KERN_INFO + "Core driver edition %02d.%02d : %s driver edition %02d.%02d\n", + DMASOUND_CORE_REVISION, DMASOUND_CORE_EDITION, dmasound.mach.name2, + (dmasound.mach.version >> 8), (dmasound.mach.version & 0xff)) ; + printk(KERN_INFO "Write will use %4d fragments of %7d bytes as default\n", + numWriteBufs, writeBufSize) ; +#ifdef HAS_RECORD + if (dmasound.mach.record) + printk(KERN_INFO + "Read will use %4d fragments of %7d bytes as default\n", + numReadBufs, readBufSize) ; +#endif + + return 0; +} + +#ifdef MODULE + +void dmasound_deinit(void) +{ + if (irq_installed) { + sound_silence(); + dmasound.mach.irqcleanup(); + irq_installed = 0; + } + + write_sq_release_buffers(); + read_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); +} + +#else /* !MODULE */ + +static int dmasound_setup(char *str) +{ + int ints[6], size; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + /* check the bootstrap parameter for "dmasound=" */ + + /* FIXME: other than in the most naive of cases there is no sense in these + * buffers being other than powers of two. This is not checked yet. + */ + + switch (ints[0]) { +#ifdef HAS_RECORD + case 5: + if ((ints[5] < 0) || (ints[5] > MAX_CATCH_RADIUS)) + printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius); + else + catchRadius = ints[5]; + /* fall through */ + case 4: + if (ints[4] < MIN_BUFFERS) + printk("dmasound_setup: invalid number of read buffers, using default = %d\n", + numReadBufs); + else + numReadBufs = ints[4]; + /* fall through */ + case 3: + if ((size = ints[3]) < 256) /* check for small buffer specs */ + size <<= 10 ; + if (size < MIN_BUFSIZE || size > MAX_BUFSIZE) + printk("dmasound_setup: invalid read buffer size, using default = %d\n", readBufSize); + else + readBufSize = size; + /* fall through */ +#else + 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 */ +#endif + case 2: + if (ints[1] < MIN_BUFFERS) + printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs); + else + numWriteBufs = ints[1]; + /* fall through */ + case 1: + if ((size = ints[2]) < 256) /* check for small buffer specs */ + size <<= 10 ; + if (size < MIN_BUFSIZE || size > MAX_BUFSIZE) + printk("dmasound_setup: invalid write buffer size, using default = %d\n", writeBufSize); + else + writeBufSize = size; + case 0: + break; + default: + printk("dmasound_setup: invalid number of arguments\n"); + return 0; + } + return 1; +} + +__setup("dmasound=", dmasound_setup); + +#endif /* !MODULE */ + + /* + * Conversion tables + */ + +#ifdef HAS_8BIT_TABLES +/* 8 bit mu-law */ + +char dmasound_ulaw2dma8[] = { + -126, -122, -118, -114, -110, -106, -102, -98, + -94, -90, -86, -82, -78, -74, -70, -66, + -63, -61, -59, -57, -55, -53, -51, -49, + -47, -45, -43, -41, -39, -37, -35, -33, + -31, -30, -29, -28, -27, -26, -25, -24, + -23, -22, -21, -20, -19, -18, -17, -16, + -16, -15, -15, -14, -14, -13, -13, -12, + -12, -11, -11, -10, -10, -9, -9, -8, + -8, -8, -7, -7, -7, -7, -6, -6, + -6, -6, -5, -5, -5, -5, -4, -4, + -4, -4, -4, -4, -3, -3, -3, -3, + -3, -3, -3, -3, -2, -2, -2, -2, + -2, -2, -2, -2, -2, -2, -2, -2, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, 0, + 125, 121, 117, 113, 109, 105, 101, 97, + 93, 89, 85, 81, 77, 73, 69, 65, + 62, 60, 58, 56, 54, 52, 50, 48, + 46, 44, 42, 40, 38, 36, 34, 32, + 30, 29, 28, 27, 26, 25, 24, 23, + 22, 21, 20, 19, 18, 17, 16, 15, + 15, 14, 14, 13, 13, 12, 12, 11, + 11, 10, 10, 9, 9, 8, 8, 7, + 7, 7, 6, 6, 6, 6, 5, 5, + 5, 5, 4, 4, 4, 4, 3, 3, + 3, 3, 3, 3, 2, 2, 2, 2, + 2, 2, 2, 2, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0 +}; + +/* 8 bit A-law */ + +char dmasound_alaw2dma8[] = { + -22, -21, -24, -23, -18, -17, -20, -19, + -30, -29, -32, -31, -26, -25, -28, -27, + -11, -11, -12, -12, -9, -9, -10, -10, + -15, -15, -16, -16, -13, -13, -14, -14, + -86, -82, -94, -90, -70, -66, -78, -74, + -118, -114, -126, -122, -102, -98, -110, -106, + -43, -41, -47, -45, -35, -33, -39, -37, + -59, -57, -63, -61, -51, -49, -55, -53, + -2, -2, -2, -2, -2, -2, -2, -2, + -2, -2, -2, -2, -2, -2, -2, -2, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -6, -6, -6, -6, -5, -5, -5, -5, + -8, -8, -8, -8, -7, -7, -7, -7, + -3, -3, -3, -3, -3, -3, -3, -3, + -4, -4, -4, -4, -4, -4, -4, -4, + 21, 20, 23, 22, 17, 16, 19, 18, + 29, 28, 31, 30, 25, 24, 27, 26, + 10, 10, 11, 11, 8, 8, 9, 9, + 14, 14, 15, 15, 12, 12, 13, 13, + 86, 82, 94, 90, 70, 66, 78, 74, + 118, 114, 126, 122, 102, 98, 110, 106, + 43, 41, 47, 45, 35, 33, 39, 37, + 59, 57, 63, 61, 51, 49, 55, 53, + 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 5, 5, 5, 5, 4, 4, 4, 4, + 7, 7, 7, 7, 6, 6, 6, 6, + 2, 2, 2, 2, 2, 2, 2, 2, + 3, 3, 3, 3, 3, 3, 3, 3 +}; +#endif /* HAS_8BIT_TABLES */ + + /* + * Visible symbols for modules + */ + +EXPORT_SYMBOL(dmasound); +EXPORT_SYMBOL(dmasound_init); +#ifdef MODULE +EXPORT_SYMBOL(dmasound_deinit); +#endif +EXPORT_SYMBOL(dmasound_write_sq); +#ifdef HAS_RECORD +EXPORT_SYMBOL(dmasound_read_sq); +#endif +EXPORT_SYMBOL(dmasound_catchRadius); +#ifdef HAS_8BIT_TABLES +EXPORT_SYMBOL(dmasound_ulaw2dma8); +EXPORT_SYMBOL(dmasound_alaw2dma8); +#endif +EXPORT_SYMBOL(get_afmt_string) ; diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c new file mode 100644 index 0000000..558db53 --- /dev/null +++ b/sound/oss/dmasound/dmasound_paula.c @@ -0,0 +1,743 @@ +/* + * linux/sound/oss/dmasound/dmasound_paula.c + * + * Amiga `Paula' DMA Sound Driver + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits + * prior to 28/01/2001 + * + * 28/01/2001 [0.1] Iain Sandoe + * - added versioning + * - put in and populated the hardware_afmts field. + * [0.2] - put in SNDCTL_DSP_GETCAPS value. + * [0.3] - put in constraint on state buffer usage. + * [0.4] - put in default hard/soft settings +*/ + + +#include <linux/module.h> +#include <linux/config.h> +#include <linux/mm.h> +#include <linux/init.h> +#include <linux/ioport.h> +#include <linux/soundcard.h> +#include <linux/interrupt.h> + +#include <asm/uaccess.h> +#include <asm/setup.h> +#include <asm/amigahw.h> +#include <asm/amigaints.h> +#include <asm/machdep.h> + +#include "dmasound.h" + +#define DMASOUND_PAULA_REVISION 0 +#define DMASOUND_PAULA_EDITION 4 + + /* + * The minimum period for audio depends on htotal (for OCS/ECS/AGA) + * (Imported from arch/m68k/amiga/amisound.c) + */ + +extern volatile u_short amiga_audio_min_period; + + + /* + * amiga_mksound() should be able to restore the period after beeping + * (Imported from arch/m68k/amiga/amisound.c) + */ + +extern u_short amiga_audio_period; + + + /* + * Audio DMA masks + */ + +#define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3) +#define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1) +#define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3) + + + /* + * Helper pointers for 16(14)-bit sound + */ + +static int write_sq_block_size_half, write_sq_block_size_quarter; + + +/*** Low level stuff *********************************************************/ + + +static void *AmiAlloc(unsigned int size, int flags); +static void AmiFree(void *obj, unsigned int size); +static int AmiIrqInit(void); +#ifdef MODULE +static void AmiIrqCleanUp(void); +#endif +static void AmiSilence(void); +static void AmiInit(void); +static int AmiSetFormat(int format); +static int AmiSetVolume(int volume); +static int AmiSetTreble(int treble); +static void AmiPlayNextFrame(int index); +static void AmiPlay(void); +static irqreturn_t AmiInterrupt(int irq, void *dummy, struct pt_regs *fp); + +#ifdef CONFIG_HEARTBEAT + + /* + * Heartbeat interferes with sound since the 7 kHz low-pass filter and the + * power LED are controlled by the same line. + */ + +#ifdef CONFIG_APUS +#define mach_heartbeat ppc_md.heartbeat +#endif + +static void (*saved_heartbeat)(int) = NULL; + +static inline void disable_heartbeat(void) +{ + if (mach_heartbeat) { + saved_heartbeat = mach_heartbeat; + mach_heartbeat = NULL; + } + AmiSetTreble(dmasound.treble); +} + +static inline void enable_heartbeat(void) +{ + if (saved_heartbeat) + mach_heartbeat = saved_heartbeat; +} +#else /* !CONFIG_HEARTBEAT */ +#define disable_heartbeat() do { } while (0) +#define enable_heartbeat() do { } while (0) +#endif /* !CONFIG_HEARTBEAT */ + + +/*** Mid level stuff *********************************************************/ + +static void AmiMixerInit(void); +static int AmiMixerIoctl(u_int cmd, u_long arg); +static int AmiWriteSqSetup(void); +static int AmiStateInfo(char *buffer, size_t space); + + +/*** 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) + */ + + + /* + * Native format + */ + +static ssize_t ami_ct_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + void *p = &frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft) & ~1; + used = count; + if (copy_from_user(p, userPtr, count)) + return -EFAULT; + } else { + u_char *left = &frame[*frameUsed>>1]; + u_char *right = left+write_sq_block_size_half; + count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1; + used = count*2; + while (count > 0) { + if (get_user(*left++, userPtr++) + || get_user(*right++, userPtr++)) + return -EFAULT; + count--; + } + } + *frameUsed += used; + return used; +} + + + /* + * Copy and convert 8 bit data + */ + +#define GENERATE_AMI_CT8(funcname, convsample) \ +static ssize_t funcname(const u_char *userPtr, size_t userCount, \ + u_char frame[], ssize_t *frameUsed, \ + ssize_t frameLeft) \ +{ \ + ssize_t count, used; \ + \ + if (!dmasound.soft.stereo) { \ + u_char *p = &frame[*frameUsed]; \ + count = min_t(size_t, userCount, frameLeft) & ~1; \ + used = count; \ + while (count > 0) { \ + u_char data; \ + if (get_user(data, userPtr++)) \ + return -EFAULT; \ + *p++ = convsample(data); \ + count--; \ + } \ + } else { \ + u_char *left = &frame[*frameUsed>>1]; \ + u_char *right = left+write_sq_block_size_half; \ + count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \ + used = count*2; \ + while (count > 0) { \ + u_char data; \ + if (get_user(data, userPtr++)) \ + return -EFAULT; \ + *left++ = convsample(data); \ + if (get_user(data, userPtr++)) \ + return -EFAULT; \ + *right++ = convsample(data); \ + count--; \ + } \ + } \ + *frameUsed += used; \ + return used; \ +} + +#define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)]) +#define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)]) +#define AMI_CT_U8(x) ((x) ^ 0x80) + +GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW) +GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW) +GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8) + + + /* + * Copy and convert 16 bit data + */ + +#define GENERATE_AMI_CT_16(funcname, convsample) \ +static ssize_t funcname(const u_char *userPtr, size_t userCount, \ + u_char frame[], ssize_t *frameUsed, \ + ssize_t frameLeft) \ +{ \ + ssize_t count, used; \ + u_short data; \ + \ + if (!dmasound.soft.stereo) { \ + u_char *high = &frame[*frameUsed>>1]; \ + u_char *low = high+write_sq_block_size_half; \ + count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \ + used = count*2; \ + while (count > 0) { \ + if (get_user(data, ((u_short *)userPtr)++)) \ + return -EFAULT; \ + data = convsample(data); \ + *high++ = data>>8; \ + *low++ = (data>>2) & 0x3f; \ + count--; \ + } \ + } else { \ + u_char *lefth = &frame[*frameUsed>>2]; \ + u_char *leftl = lefth+write_sq_block_size_quarter; \ + u_char *righth = lefth+write_sq_block_size_half; \ + u_char *rightl = righth+write_sq_block_size_quarter; \ + count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \ + used = count*4; \ + while (count > 0) { \ + if (get_user(data, ((u_short *)userPtr)++)) \ + return -EFAULT; \ + data = convsample(data); \ + *lefth++ = data>>8; \ + *leftl++ = (data>>2) & 0x3f; \ + if (get_user(data, ((u_short *)userPtr)++)) \ + return -EFAULT; \ + data = convsample(data); \ + *righth++ = data>>8; \ + *rightl++ = (data>>2) & 0x3f; \ + count--; \ + } \ + } \ + *frameUsed += used; \ + return used; \ +} + +#define AMI_CT_S16BE(x) (x) +#define AMI_CT_U16BE(x) ((x) ^ 0x8000) +#define AMI_CT_S16LE(x) (le2be16((x))) +#define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000) + +GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE) +GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE) +GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE) +GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE) + + +static TRANS transAmiga = { + .ct_ulaw = ami_ct_ulaw, + .ct_alaw = ami_ct_alaw, + .ct_s8 = ami_ct_s8, + .ct_u8 = ami_ct_u8, + .ct_s16be = ami_ct_s16be, + .ct_u16be = ami_ct_u16be, + .ct_s16le = ami_ct_s16le, + .ct_u16le = ami_ct_u16le, +}; + +/*** Low level stuff *********************************************************/ + +static inline void StopDMA(void) +{ + custom.aud[0].audvol = custom.aud[1].audvol = 0; + custom.aud[2].audvol = custom.aud[3].audvol = 0; + custom.dmacon = AMI_AUDIO_OFF; + enable_heartbeat(); +} + +static void *AmiAlloc(unsigned int size, int flags) +{ + return amiga_chip_alloc((long)size, "dmasound [Paula]"); +} + +static void AmiFree(void *obj, unsigned int size) +{ + amiga_chip_free (obj); +} + +static int __init AmiIrqInit(void) +{ + /* turn off DMA for audio channels */ + StopDMA(); + + /* Register interrupt handler. */ + if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound", + AmiInterrupt)) + return 0; + return 1; +} + +#ifdef MODULE +static void AmiIrqCleanUp(void) +{ + /* turn off DMA for audio channels */ + StopDMA(); + /* release the interrupt */ + free_irq(IRQ_AMIGA_AUD0, AmiInterrupt); +} +#endif /* MODULE */ + +static void AmiSilence(void) +{ + /* turn off DMA for audio channels */ + StopDMA(); +} + + +static void AmiInit(void) +{ + int period, i; + + AmiSilence(); + + if (dmasound.soft.speed) + period = amiga_colorclock/dmasound.soft.speed-1; + else + period = amiga_audio_min_period; + dmasound.hard = dmasound.soft; + dmasound.trans_write = &transAmiga; + + if (period < amiga_audio_min_period) { + /* we would need to squeeze the sound, but we won't do that */ + period = amiga_audio_min_period; + } else if (period > 65535) { + period = 65535; + } + dmasound.hard.speed = amiga_colorclock/(period+1); + + for (i = 0; i < 4; i++) + custom.aud[i].audper = period; + amiga_audio_period = period; +} + + +static int AmiSetFormat(int format) +{ + int size; + + /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */ + + switch (format) { + case AFMT_QUERY: + return dmasound.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: /* :-) */ + size = 8; + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = size; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = dmasound.soft.size; + } + AmiInit(); + + return format; +} + + +#define VOLUME_VOXWARE_TO_AMI(v) \ + (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100) +#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64) + +static int AmiSetVolume(int volume) +{ + dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff); + custom.aud[0].audvol = dmasound.volume_left; + dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8); + custom.aud[1].audvol = dmasound.volume_right; + if (dmasound.hard.size == 16) { + if (dmasound.volume_left == 64 && dmasound.volume_right == 64) { + custom.aud[2].audvol = 1; + custom.aud[3].audvol = 1; + } else { + custom.aud[2].audvol = 0; + custom.aud[3].audvol = 0; + } + } + return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) | + (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8); +} + +static int AmiSetTreble(int treble) +{ + dmasound.treble = treble; + if (treble < 50) + ciaa.pra &= ~0x02; + else + ciaa.pra |= 0x02; + return treble; +} + + +#define AMI_PLAY_LOADED 1 +#define AMI_PLAY_PLAYING 2 +#define AMI_PLAY_MASK 3 + + +static void AmiPlayNextFrame(int index) +{ + u_char *start, *ch0, *ch1, *ch2, *ch3; + u_long size; + + /* used by AmiPlay() if all doubts whether there really is something + * to be played are already wiped out. + */ + start = write_sq.buffers[write_sq.front]; + size = (write_sq.count == index ? write_sq.rear_size + : write_sq.block_size)>>1; + + if (dmasound.hard.stereo) { + ch0 = start; + ch1 = start+write_sq_block_size_half; + size >>= 1; + } else { + ch0 = start; + ch1 = start; + } + + disable_heartbeat(); + custom.aud[0].audvol = dmasound.volume_left; + custom.aud[1].audvol = dmasound.volume_right; + if (dmasound.hard.size == 8) { + custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0); + custom.aud[0].audlen = size; + custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1); + custom.aud[1].audlen = size; + custom.dmacon = AMI_AUDIO_8; + } else { + size >>= 1; + custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0); + custom.aud[0].audlen = size; + custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1); + custom.aud[1].audlen = size; + if (dmasound.volume_left == 64 && dmasound.volume_right == 64) { + /* We can play pseudo 14-bit only with the maximum volume */ + ch3 = ch0+write_sq_block_size_quarter; + ch2 = ch1+write_sq_block_size_quarter; + custom.aud[2].audvol = 1; /* we are being affected by the beeps */ + custom.aud[3].audvol = 1; /* restoring volume here helps a bit */ + custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2); + custom.aud[2].audlen = size; + custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3); + custom.aud[3].audlen = size; + custom.dmacon = AMI_AUDIO_14; + } else { + custom.aud[2].audvol = 0; + custom.aud[3].audvol = 0; + custom.dmacon = AMI_AUDIO_8; + } + } + write_sq.front = (write_sq.front+1) % write_sq.max_count; + write_sq.active |= AMI_PLAY_LOADED; +} + + +static void AmiPlay(void) +{ + int minframes = 1; + + custom.intena = IF_AUD0; + + if (write_sq.active & AMI_PLAY_LOADED) { + /* There's already a frame loaded */ + custom.intena = IF_SETCLR | IF_AUD0; + return; + } + + if (write_sq.active & AMI_PLAY_PLAYING) + /* Increase threshold: frame 1 is already being played */ + minframes = 2; + + if (write_sq.count < minframes) { + /* Nothing to do */ + custom.intena = IF_SETCLR | IF_AUD0; + return; + } + + if (write_sq.count <= minframes && + write_sq.rear_size < write_sq.block_size && !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + custom.intena = IF_SETCLR | IF_AUD0; + return; + } + + AmiPlayNextFrame(minframes); + + custom.intena = IF_SETCLR | IF_AUD0; +} + + +static irqreturn_t AmiInterrupt(int irq, void *dummy, struct pt_regs *fp) +{ + int minframes = 1; + + custom.intena = IF_AUD0; + + if (!write_sq.active) { + /* Playing was interrupted and sq_reset() has already cleared + * the sq variables, so better don't do anything here. + */ + WAKE_UP(write_sq.sync_queue); + return IRQ_HANDLED; + } + + if (write_sq.active & AMI_PLAY_PLAYING) { + /* We've just finished a frame */ + write_sq.count--; + WAKE_UP(write_sq.action_queue); + } + + if (write_sq.active & AMI_PLAY_LOADED) + /* Increase threshold: frame 1 is already being played */ + minframes = 2; + + /* Shift the flags */ + write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK; + + if (!write_sq.active) + /* No frame is playing, disable audio DMA */ + StopDMA(); + + custom.intena = IF_SETCLR | IF_AUD0; + + if (write_sq.count >= minframes) + /* Try to play the next frame */ + AmiPlay(); + + if (!write_sq.active) + /* Nothing to play anymore. + Wake up a process waiting for audio output to drain. */ + WAKE_UP(write_sq.sync_queue); + return IRQ_HANDLED; +} + +/*** Mid level stuff *********************************************************/ + + +/* + * /dev/mixer abstraction + */ + +static void __init AmiMixerInit(void) +{ + dmasound.volume_left = 64; + dmasound.volume_right = 64; + custom.aud[0].audvol = dmasound.volume_left; + custom.aud[3].audvol = 1; /* For pseudo 14bit */ + custom.aud[1].audvol = dmasound.volume_right; + custom.aud[2].audvol = 1; /* For pseudo 14bit */ + dmasound.treble = 50; +} + +static int AmiMixerIoctl(u_int cmd, u_long arg) +{ + int data; + switch (cmd) { + case SOUND_MIXER_READ_DEVMASK: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE); + case SOUND_MIXER_READ_RECMASK: + return IOCTL_OUT(arg, 0); + case SOUND_MIXER_READ_STEREODEVS: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME); + case SOUND_MIXER_READ_VOLUME: + return IOCTL_OUT(arg, + VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) | + VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8); + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_volume(data)); + case SOUND_MIXER_READ_TREBLE: + return IOCTL_OUT(arg, dmasound.treble); + case SOUND_MIXER_WRITE_TREBLE: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_treble(data)); + } + return -EINVAL; +} + + +static int AmiWriteSqSetup(void) +{ + write_sq_block_size_half = write_sq.block_size>>1; + write_sq_block_size_quarter = write_sq_block_size_half>>1; + return 0; +} + + +static int AmiStateInfo(char *buffer, size_t space) +{ + int len = 0; + len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n", + dmasound.volume_left); + len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n", + dmasound.volume_right); + if (len >= space) { + printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ; + len = space ; + } + return len; +} + + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard = { + .format = AFMT_S8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static SETTINGS def_soft = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static MACHINE machAmiga = { + .name = "Amiga", + .name2 = "AMIGA", + .owner = THIS_MODULE, + .dma_alloc = AmiAlloc, + .dma_free = AmiFree, + .irqinit = AmiIrqInit, +#ifdef MODULE + .irqcleanup = AmiIrqCleanUp, +#endif /* MODULE */ + .init = AmiInit, + .silence = AmiSilence, + .setFormat = AmiSetFormat, + .setVolume = AmiSetVolume, + .setTreble = AmiSetTreble, + .play = AmiPlay, + .mixer_init = AmiMixerInit, + .mixer_ioctl = AmiMixerIoctl, + .write_sq_setup = AmiWriteSqSetup, + .state_info = AmiStateInfo, + .min_dsp_speed = 8000, + .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION), + .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + + +/*** Config & Setup **********************************************************/ + + +int __init dmasound_paula_init(void) +{ + int err; + + if (MACH_IS_AMIGA && AMIGAHW_PRESENT(AMI_AUDIO)) { + if (!request_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40, + "dmasound [Paula]")) + return -EBUSY; + dmasound.mach = machAmiga; + dmasound.mach.default_hard = def_hard ; + dmasound.mach.default_soft = def_soft ; + err = dmasound_init(); + if (err) + release_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40); + return err; + } else + return -ENODEV; +} + +static void __exit dmasound_paula_cleanup(void) +{ + dmasound_deinit(); + release_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40); +} + +module_init(dmasound_paula_init); +module_exit(dmasound_paula_cleanup); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c new file mode 100644 index 0000000..92c25a0 --- /dev/null +++ b/sound/oss/dmasound/dmasound_q40.c @@ -0,0 +1,634 @@ +/* + * linux/sound/oss/dmasound/dmasound_q40.c + * + * Q40 DMA Sound Driver + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits + * prior to 28/01/2001 + * + * 28/01/2001 [0.1] Iain Sandoe + * - added versioning + * - put in and populated the hardware_afmts field. + * [0.2] - put in SNDCTL_DSP_GETCAPS value. + * [0.3] - put in default hard/soft settings. + */ + + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/soundcard.h> +#include <linux/interrupt.h> + +#include <asm/uaccess.h> +#include <asm/q40ints.h> +#include <asm/q40_master.h> + +#include "dmasound.h" + +#define DMASOUND_Q40_REVISION 0 +#define DMASOUND_Q40_EDITION 3 + +static int expand_bal; /* Balance factor for expanding (not volume!) */ +static int expand_data; /* Data for expanding */ + + +/*** Low level stuff *********************************************************/ + + +static void *Q40Alloc(unsigned int size, int flags); +static void Q40Free(void *, unsigned int); +static int Q40IrqInit(void); +#ifdef MODULE +static void Q40IrqCleanUp(void); +#endif +static void Q40Silence(void); +static void Q40Init(void); +static int Q40SetFormat(int format); +static int Q40SetVolume(int volume); +static void Q40PlayNextFrame(int index); +static void Q40Play(void); +static irqreturn_t Q40StereoInterrupt(int irq, void *dummy, struct pt_regs *fp); +static irqreturn_t Q40MonoInterrupt(int irq, void *dummy, struct pt_regs *fp); +static void Q40Interrupt(void); + + +/*** Mid level stuff *********************************************************/ + + + +/* userCount, frameUsed, frameLeft == byte counts */ +static ssize_t q40_ct_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8; + ssize_t count, used; + u_char *p = (u_char *) &frame[*frameUsed]; + + used = count = min_t(size_t, userCount, frameLeft); + if (copy_from_user(p,userPtr,count)) + return -EFAULT; + while (count > 0) { + *p = table[*p]+128; + p++; + count--; + } + *frameUsed += used ; + return used; +} + + +static ssize_t q40_ct_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + u_char *p = (u_char *) &frame[*frameUsed]; + + used = count = min_t(size_t, userCount, frameLeft); + if (copy_from_user(p,userPtr,count)) + return -EFAULT; + while (count > 0) { + *p = *p + 128; + p++; + count--; + } + *frameUsed += used; + return used; +} + +static ssize_t q40_ct_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + u_char *p = (u_char *) &frame[*frameUsed]; + + used = count = min_t(size_t, userCount, frameLeft); + if (copy_from_user(p,userPtr,count)) + return -EFAULT; + *frameUsed += used; + return used; +} + + +/* a bit too complicated to optimise right now ..*/ +static ssize_t q40_ctx_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + unsigned char *table = (unsigned char *) + (dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8); + unsigned int data = expand_data; + u_char *p = (u_char *) &frame[*frameUsed]; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + 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]; + data += 0x80; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctx_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + + 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 ; + data += 0x80; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctx_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + 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 ; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft) ; + utotal -= userCount; + return utotal; +} + +/* compressing versions */ +static ssize_t q40_ctc_law(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + unsigned char *table = (unsigned char *) + (dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8); + unsigned int data = expand_data; + u_char *p = (u_char *) &frame[*frameUsed]; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + while(bal<0) { + if (userCount == 0) + goto lout; + if (!(bal<(-hSpeed))) { + if (get_user(c, userPtr)) + return -EFAULT; + data = 0x80 + table[c]; + } + userPtr++; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + lout: + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctc_s8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + while (bal < 0) { + if (userCount == 0) + goto lout; + if (!(bal<(-hSpeed))) { + if (get_user(c, userPtr)) + return -EFAULT; + data = c + 0x80; + } + userPtr++; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + lout: + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctc_u8(const u_char *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + while (bal < 0) { + if (userCount == 0) + goto lout; + if (!(bal<(-hSpeed))) { + if (get_user(c, userPtr)) + return -EFAULT; + data = c ; + } + userPtr++; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + lout: + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft) ; + utotal -= userCount; + return utotal; +} + + +static TRANS transQ40Normal = { + q40_ct_law, q40_ct_law, q40_ct_s8, q40_ct_u8, NULL, NULL, NULL, NULL +}; + +static TRANS transQ40Expanding = { + q40_ctx_law, q40_ctx_law, q40_ctx_s8, q40_ctx_u8, NULL, NULL, NULL, NULL +}; + +static TRANS transQ40Compressing = { + q40_ctc_law, q40_ctc_law, q40_ctc_s8, q40_ctc_u8, NULL, NULL, NULL, NULL +}; + + +/*** Low level stuff *********************************************************/ + +static void *Q40Alloc(unsigned int size, int flags) +{ + return kmalloc(size, flags); /* change to vmalloc */ +} + +static void Q40Free(void *ptr, unsigned int size) +{ + kfree(ptr); +} + +static int __init Q40IrqInit(void) +{ + /* Register interrupt handler. */ + request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "DMA sound", Q40Interrupt); + + return(1); +} + + +#ifdef MODULE +static void Q40IrqCleanUp(void) +{ + master_outb(0,SAMPLE_ENABLE_REG); + free_irq(Q40_IRQ_SAMPLE, Q40Interrupt); +} +#endif /* MODULE */ + + +static void Q40Silence(void) +{ + master_outb(0,SAMPLE_ENABLE_REG); + *DAC_LEFT=*DAC_RIGHT=127; +} + +static char *q40_pp; +static unsigned int q40_sc; + +static void Q40PlayNextFrame(int index) +{ + u_char *start; + u_long size; + u_char speed; + + /* used by Q40Play() if all doubts whether there really is something + * to be played are already wiped out. + */ + start = write_sq.buffers[write_sq.front]; + size = (write_sq.count == index ? write_sq.rear_size : write_sq.block_size); + + q40_pp=start; + q40_sc=size; + + write_sq.front = (write_sq.front+1) % write_sq.max_count; + write_sq.active++; + + speed=(dmasound.hard.speed==10000 ? 0 : 1); + + master_outb( 0,SAMPLE_ENABLE_REG); + free_irq(Q40_IRQ_SAMPLE, Q40Interrupt); + if (dmasound.soft.stereo) + request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "Q40 sound", Q40Interrupt); + else + request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0, + "Q40 sound", Q40Interrupt); + + master_outb( speed, SAMPLE_RATE_REG); + master_outb( 1,SAMPLE_CLEAR_REG); + master_outb( 1,SAMPLE_ENABLE_REG); +} + +static void Q40Play(void) +{ + unsigned long flags; + + if (write_sq.active || write_sq.count<=0 ) { + /* There's already a frame loaded */ + return; + } + + /* nothing in the queue */ + if (write_sq.count <= 1 && write_sq.rear_size < write_sq.block_size && !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + return; + } + spin_lock_irqsave(&dmasound.lock, flags); + Q40PlayNextFrame(1); + spin_unlock_irqrestore(&dmasound.lock, flags); +} + +static irqreturn_t Q40StereoInterrupt(int irq, void *dummy, struct pt_regs *fp) +{ + spin_lock(&dmasound.lock); + if (q40_sc>1){ + *DAC_LEFT=*q40_pp++; + *DAC_RIGHT=*q40_pp++; + q40_sc -=2; + master_outb(1,SAMPLE_CLEAR_REG); + }else Q40Interrupt(); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} +static irqreturn_t Q40MonoInterrupt(int irq, void *dummy, struct pt_regs *fp) +{ + spin_lock(&dmasound.lock); + if (q40_sc>0){ + *DAC_LEFT=*q40_pp; + *DAC_RIGHT=*q40_pp++; + q40_sc --; + master_outb(1,SAMPLE_CLEAR_REG); + }else Q40Interrupt(); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} +static void Q40Interrupt(void) +{ + if (!write_sq.active) { + /* playing was interrupted and sq_reset() has already cleared + * the sq variables, so better don't do anything here. + */ + WAKE_UP(write_sq.sync_queue); + master_outb(0,SAMPLE_ENABLE_REG); /* better safe */ + goto exit; + } else write_sq.active=0; + write_sq.count--; + Q40Play(); + + if (q40_sc<2) + { /* there was nothing to play, disable irq */ + master_outb(0,SAMPLE_ENABLE_REG); + *DAC_LEFT=*DAC_RIGHT=127; + } + WAKE_UP(write_sq.action_queue); + + exit: + master_outb(1,SAMPLE_CLEAR_REG); +} + + +static void Q40Init(void) +{ + int i, idx; + const int freq[] = {10000, 20000}; + + /* search a frequency that fits into the allowed error range */ + + idx = -1; + for (i = 0; i < 2; i++) + if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) <= catchRadius) + idx = i; + + dmasound.hard = dmasound.soft; + /*sound.hard.stereo=1;*/ /* no longer true */ + dmasound.hard.size=8; + + if (idx > -1) { + dmasound.soft.speed = freq[idx]; + dmasound.trans_write = &transQ40Normal; + } else + dmasound.trans_write = &transQ40Expanding; + + Q40Silence(); + + if (dmasound.hard.speed > 20200) { + /* squeeze the sound, we do that */ + dmasound.hard.speed = 20000; + dmasound.trans_write = &transQ40Compressing; + } else if (dmasound.hard.speed > 10000) { + dmasound.hard.speed = 20000; + } else { + dmasound.hard.speed = 10000; + } + expand_bal = -dmasound.soft.speed; +} + + +static int Q40SetFormat(int format) +{ + /* Q40 sound supports only 8bit modes */ + + switch (format) { + case AFMT_QUERY: + return(dmasound.soft.format); + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_S8: + case AFMT_U8: + break; + default: + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = 8; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = 8; + } + Q40Init(); + + return(format); +} + +static int Q40SetVolume(int volume) +{ + return 0; +} + + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 10000 +} ; + +static SETTINGS def_soft = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static MACHINE machQ40 = { + .name = "Q40", + .name2 = "Q40", + .owner = THIS_MODULE, + .dma_alloc = Q40Alloc, + .dma_free = Q40Free, + .irqinit = Q40IrqInit, +#ifdef MODULE + .irqcleanup = Q40IrqCleanUp, +#endif /* MODULE */ + .init = Q40Init, + .silence = Q40Silence, + .setFormat = Q40SetFormat, + .setVolume = Q40SetVolume, + .play = Q40Play, + .min_dsp_speed = 10000, + .version = ((DMASOUND_Q40_REVISION<<8) | DMASOUND_Q40_EDITION), + .hardware_afmts = AFMT_U8, /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + + +/*** Config & Setup **********************************************************/ + + +int __init dmasound_q40_init(void) +{ + if (MACH_IS_Q40) { + dmasound.mach = machQ40; + dmasound.mach.default_hard = def_hard ; + dmasound.mach.default_soft = def_soft ; + return dmasound_init(); + } else + return -ENODEV; +} + +static void __exit dmasound_q40_cleanup(void) +{ + dmasound_deinit(); +} + +module_init(dmasound_q40_init); +module_exit(dmasound_q40_cleanup); + +MODULE_DESCRIPTION("Q40/Q60 sound driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/dmasound/tas3001c.c b/sound/oss/dmasound/tas3001c.c new file mode 100644 index 0000000..f227c9f --- /dev/null +++ b/sound/oss/dmasound/tas3001c.c @@ -0,0 +1,850 @@ +/* + * Driver for the i2c/i2s based TA3004 sound chip used + * on some Apple hardware. Also known as "snapper". + * + * Tobias Sargeant <tobias.sargeant@bigpond.com> + * Based upon, tas3001c.c by Christopher C. Chimelis <chris@debian.org>: + * + * TODO: + * ----- + * * Enable control over input line 2 (is this connected?) + * * Implement sleep support (at least mute everything and + * * set gains to minimum during sleep) + * * Look into some of Darwin's tweaks regarding the mute + * * lines (delays & different behaviour on some HW) + * + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/proc_fs.h> +#include <linux/ioport.h> +#include <linux/sysctl.h> +#include <linux/types.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <linux/workqueue.h> +#include <asm/uaccess.h> +#include <asm/errno.h> +#include <asm/io.h> +#include <asm/prom.h> + +#include "dmasound.h" +#include "tas_common.h" +#include "tas3001c.h" + +#include "tas_ioctl.h" + +#define TAS3001C_BIQUAD_FILTER_COUNT 6 +#define TAS3001C_BIQUAD_CHANNEL_COUNT 2 + +#define VOL_DEFAULT (100 * 4 / 5) +#define INPUT_DEFAULT (100 * 4 / 5) +#define BASS_DEFAULT (100 / 2) +#define TREBLE_DEFAULT (100 / 2) + +struct tas3001c_data_t { + struct tas_data_t super; + int device_id; + int output_id; + int speaker_id; + struct tas_drce_t drce_state; +}; + + +static const union tas_biquad_t +tas3001c_eq_unity={ + .buf = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } +}; + + +static inline unsigned char db_to_regval(short db) { + int r=0; + + r=(db+0x59a0) / 0x60; + + if (r < 0x91) return 0x91; + if (r > 0xef) return 0xef; + return r; +} + +static inline short quantize_db(short db) { + return db_to_regval(db) * 0x60 - 0x59a0; +} + + +static inline int +register_width(enum tas3001c_reg_t r) +{ + switch(r) { + case TAS3001C_REG_MCR: + case TAS3001C_REG_TREBLE: + case TAS3001C_REG_BASS: + return 1; + + case TAS3001C_REG_DRC: + return 2; + + case TAS3001C_REG_MIXER1: + case TAS3001C_REG_MIXER2: + return 3; + + case TAS3001C_REG_VOLUME: + return 6; + + case TAS3001C_REG_LEFT_BIQUAD0: + case TAS3001C_REG_LEFT_BIQUAD1: + case TAS3001C_REG_LEFT_BIQUAD2: + case TAS3001C_REG_LEFT_BIQUAD3: + case TAS3001C_REG_LEFT_BIQUAD4: + case TAS3001C_REG_LEFT_BIQUAD5: + case TAS3001C_REG_LEFT_BIQUAD6: + + case TAS3001C_REG_RIGHT_BIQUAD0: + case TAS3001C_REG_RIGHT_BIQUAD1: + case TAS3001C_REG_RIGHT_BIQUAD2: + case TAS3001C_REG_RIGHT_BIQUAD3: + case TAS3001C_REG_RIGHT_BIQUAD4: + case TAS3001C_REG_RIGHT_BIQUAD5: + case TAS3001C_REG_RIGHT_BIQUAD6: + return 15; + + default: + return 0; + } +} + +static int +tas3001c_write_register( struct tas3001c_data_t *self, + enum tas3001c_reg_t reg_num, + char *data, + uint write_mode) +{ + if (reg_num==TAS3001C_REG_MCR || + reg_num==TAS3001C_REG_BASS || + reg_num==TAS3001C_REG_TREBLE) { + return tas_write_byte_register(&self->super, + (uint)reg_num, + *data, + write_mode); + } else { + return tas_write_register(&self->super, + (uint)reg_num, + register_width(reg_num), + data, + write_mode); + } +} + +static int +tas3001c_sync_register( struct tas3001c_data_t *self, + enum tas3001c_reg_t reg_num) +{ + if (reg_num==TAS3001C_REG_MCR || + reg_num==TAS3001C_REG_BASS || + reg_num==TAS3001C_REG_TREBLE) { + return tas_sync_byte_register(&self->super, + (uint)reg_num, + register_width(reg_num)); + } else { + return tas_sync_register(&self->super, + (uint)reg_num, + register_width(reg_num)); + } +} + +static int +tas3001c_read_register( struct tas3001c_data_t *self, + enum tas3001c_reg_t reg_num, + char *data, + uint write_mode) +{ + return tas_read_register(&self->super, + (uint)reg_num, + register_width(reg_num), + data); +} + +static inline int +tas3001c_fast_load(struct tas3001c_data_t *self, int fast) +{ + if (fast) + self->super.shadow[TAS3001C_REG_MCR][0] |= 0x80; + else + self->super.shadow[TAS3001C_REG_MCR][0] &= 0x7f; + return tas3001c_sync_register(self,TAS3001C_REG_MCR); +} + +static uint +tas3001c_supported_mixers(struct tas3001c_data_t *self) +{ + return SOUND_MASK_VOLUME | + SOUND_MASK_PCM | + SOUND_MASK_ALTPCM | + SOUND_MASK_TREBLE | + SOUND_MASK_BASS; +} + +static int +tas3001c_mixer_is_stereo(struct tas3001c_data_t *self,int mixer) +{ + switch(mixer) { + case SOUND_MIXER_VOLUME: + return 1; + default: + return 0; + } +} + +static uint +tas3001c_stereo_mixers(struct tas3001c_data_t *self) +{ + uint r=tas3001c_supported_mixers(self); + uint i; + + for (i=1; i<SOUND_MIXER_NRDEVICES; i++) + if (r&(1<<i) && !tas3001c_mixer_is_stereo(self,i)) + r &= ~(1<<i); + return r; +} + +static int +tas3001c_get_mixer_level(struct tas3001c_data_t *self,int mixer,uint *level) +{ + if (!self) + return -1; + + *level=self->super.mixer[mixer]; + + return 0; +} + +static int +tas3001c_set_mixer_level(struct tas3001c_data_t *self,int mixer,uint level) +{ + int rc; + tas_shadow_t *shadow; + + uint temp; + uint offset=0; + + if (!self) + return -1; + + shadow=self->super.shadow; + + if (!tas3001c_mixer_is_stereo(self,mixer)) + level = tas_mono_to_stereo(level); + + switch(mixer) { + case SOUND_MIXER_VOLUME: + temp = tas3001c_gain.master[level&0xff]; + shadow[TAS3001C_REG_VOLUME][0] = (temp >> 16) & 0xff; + shadow[TAS3001C_REG_VOLUME][1] = (temp >> 8) & 0xff; + shadow[TAS3001C_REG_VOLUME][2] = (temp >> 0) & 0xff; + temp = tas3001c_gain.master[(level>>8)&0xff]; + shadow[TAS3001C_REG_VOLUME][3] = (temp >> 16) & 0xff; + shadow[TAS3001C_REG_VOLUME][4] = (temp >> 8) & 0xff; + shadow[TAS3001C_REG_VOLUME][5] = (temp >> 0) & 0xff; + rc = tas3001c_sync_register(self,TAS3001C_REG_VOLUME); + break; + case SOUND_MIXER_ALTPCM: + /* tas3001c_fast_load(self, 1); */ + level = tas_mono_to_stereo(level); + temp = tas3001c_gain.mixer[level&0xff]; + shadow[TAS3001C_REG_MIXER2][offset+0] = (temp >> 16) & 0xff; + shadow[TAS3001C_REG_MIXER2][offset+1] = (temp >> 8) & 0xff; + shadow[TAS3001C_REG_MIXER2][offset+2] = (temp >> 0) & 0xff; + rc = tas3001c_sync_register(self,TAS3001C_REG_MIXER2); + /* tas3001c_fast_load(self, 0); */ + break; + case SOUND_MIXER_PCM: + /* tas3001c_fast_load(self, 1); */ + level = tas_mono_to_stereo(level); + temp = tas3001c_gain.mixer[level&0xff]; + shadow[TAS3001C_REG_MIXER1][offset+0] = (temp >> 16) & 0xff; + shadow[TAS3001C_REG_MIXER1][offset+1] = (temp >> 8) & 0xff; + shadow[TAS3001C_REG_MIXER1][offset+2] = (temp >> 0) & 0xff; + rc = tas3001c_sync_register(self,TAS3001C_REG_MIXER1); + /* tas3001c_fast_load(self, 0); */ + break; + case SOUND_MIXER_TREBLE: + temp = tas3001c_gain.treble[level&0xff]; + shadow[TAS3001C_REG_TREBLE][0]=temp&0xff; + rc = tas3001c_sync_register(self,TAS3001C_REG_TREBLE); + break; + case SOUND_MIXER_BASS: + temp = tas3001c_gain.bass[level&0xff]; + shadow[TAS3001C_REG_BASS][0]=temp&0xff; + rc = tas3001c_sync_register(self,TAS3001C_REG_BASS); + break; + default: + rc = -1; + break; + } + if (rc < 0) + return rc; + self->super.mixer[mixer]=level; + return 0; +} + +static int +tas3001c_leave_sleep(struct tas3001c_data_t *self) +{ + unsigned char mcr = (1<<6)+(2<<4)+(2<<2); + + if (!self) + return -1; + + /* Make sure something answers on the i2c bus */ + if (tas3001c_write_register(self, TAS3001C_REG_MCR, &mcr, + WRITE_NORMAL|FORCE_WRITE) < 0) + return -1; + + tas3001c_fast_load(self, 1); + + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD0); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD1); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD2); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD3); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD4); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD5); + + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD0); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD1); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD2); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD3); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD4); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD5); + + tas3001c_fast_load(self, 0); + + (void)tas3001c_sync_register(self,TAS3001C_REG_BASS); + (void)tas3001c_sync_register(self,TAS3001C_REG_TREBLE); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER1); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER2); + (void)tas3001c_sync_register(self,TAS3001C_REG_VOLUME); + + return 0; +} + +static int +tas3001c_enter_sleep(struct tas3001c_data_t *self) +{ + /* Stub for now, but I have the details on low-power mode */ + if (!self) + return -1; + return 0; +} + +static int +tas3001c_sync_biquad( struct tas3001c_data_t *self, + u_int channel, + u_int filter) +{ + enum tas3001c_reg_t reg; + + if (channel >= TAS3001C_BIQUAD_CHANNEL_COUNT || + filter >= TAS3001C_BIQUAD_FILTER_COUNT) return -EINVAL; + + reg=( channel ? TAS3001C_REG_RIGHT_BIQUAD0 : TAS3001C_REG_LEFT_BIQUAD0 ) + filter; + + return tas3001c_sync_register(self,reg); +} + +static int +tas3001c_write_biquad_shadow( struct tas3001c_data_t *self, + u_int channel, + u_int filter, + const union tas_biquad_t *biquad) +{ + tas_shadow_t *shadow=self->super.shadow; + enum tas3001c_reg_t reg; + + if (channel >= TAS3001C_BIQUAD_CHANNEL_COUNT || + filter >= TAS3001C_BIQUAD_FILTER_COUNT) return -EINVAL; + + reg=( channel ? TAS3001C_REG_RIGHT_BIQUAD0 : TAS3001C_REG_LEFT_BIQUAD0 ) + filter; + + SET_4_20(shadow[reg], 0,biquad->coeff.b0); + SET_4_20(shadow[reg], 3,biquad->coeff.b1); + SET_4_20(shadow[reg], 6,biquad->coeff.b2); + SET_4_20(shadow[reg], 9,biquad->coeff.a1); + SET_4_20(shadow[reg],12,biquad->coeff.a2); + + return 0; +} + +static int +tas3001c_write_biquad( struct tas3001c_data_t *self, + u_int channel, + u_int filter, + const union tas_biquad_t *biquad) +{ + int rc; + + rc=tas3001c_write_biquad_shadow(self, channel, filter, biquad); + if (rc < 0) return rc; + + return tas3001c_sync_biquad(self, channel, filter); +} + +static int +tas3001c_write_biquad_list( struct tas3001c_data_t *self, + u_int filter_count, + u_int flags, + struct tas_biquad_ctrl_t *biquads) +{ + int i; + int rc; + + if (flags & TAS_BIQUAD_FAST_LOAD) tas3001c_fast_load(self,1); + + for (i=0; i<filter_count; i++) { + rc=tas3001c_write_biquad(self, + biquads[i].channel, + biquads[i].filter, + &biquads[i].data); + if (rc < 0) break; + } + + if (flags & TAS_BIQUAD_FAST_LOAD) { + tas3001c_fast_load(self,0); + + (void)tas3001c_sync_register(self,TAS3001C_REG_BASS); + (void)tas3001c_sync_register(self,TAS3001C_REG_TREBLE); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER1); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER2); + (void)tas3001c_sync_register(self,TAS3001C_REG_VOLUME); + } + + return rc; +} + +static int +tas3001c_read_biquad( struct tas3001c_data_t *self, + u_int channel, + u_int filter, + union tas_biquad_t *biquad) +{ + tas_shadow_t *shadow=self->super.shadow; + enum tas3001c_reg_t reg; + + if (channel >= TAS3001C_BIQUAD_CHANNEL_COUNT || + filter >= TAS3001C_BIQUAD_FILTER_COUNT) return -EINVAL; + + reg=( channel ? TAS3001C_REG_RIGHT_BIQUAD0 : TAS3001C_REG_LEFT_BIQUAD0 ) + filter; + + biquad->coeff.b0=GET_4_20(shadow[reg], 0); + biquad->coeff.b1=GET_4_20(shadow[reg], 3); + biquad->coeff.b2=GET_4_20(shadow[reg], 6); + biquad->coeff.a1=GET_4_20(shadow[reg], 9); + biquad->coeff.a2=GET_4_20(shadow[reg],12); + + return 0; +} + +static int +tas3001c_eq_rw( struct tas3001c_data_t *self, + u_int cmd, + u_long arg) +{ + int rc; + struct tas_biquad_ctrl_t biquad; + void __user *argp = (void __user *)arg; + + if (copy_from_user(&biquad, argp, sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + + if (cmd & SIOC_IN) { + rc=tas3001c_write_biquad(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + } + + if (cmd & SIOC_OUT) { + rc=tas3001c_read_biquad(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + + if (copy_to_user(argp, &biquad, sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + + } + return 0; +} + +static int +tas3001c_eq_list_rw( struct tas3001c_data_t *self, + u_int cmd, + u_long arg) +{ + int rc; + int filter_count; + int flags; + int i,j; + char sync_required[2][6]; + struct tas_biquad_ctrl_t biquad; + struct tas_biquad_ctrl_list_t __user *argp = (void __user *)arg; + + memset(sync_required,0,sizeof(sync_required)); + + if (copy_from_user(&filter_count, &argp->filter_count, sizeof(int))) + return -EFAULT; + + if (copy_from_user(&flags, &argp->flags, sizeof(int))) + return -EFAULT; + + if (cmd & SIOC_IN) { + } + + for (i=0; i < filter_count; i++) { + if (copy_from_user(&biquad, &argp->biquads[i], + sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + + if (cmd & SIOC_IN) { + sync_required[biquad.channel][biquad.filter]=1; + rc=tas3001c_write_biquad_shadow(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + } + + if (cmd & SIOC_OUT) { + rc=tas3001c_read_biquad(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + + if (copy_to_user(&argp->biquads[i], &biquad, + sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + } + } + + if (cmd & SIOC_IN) { + if (flags & TAS_BIQUAD_FAST_LOAD) tas3001c_fast_load(self,1); + for (i=0; i<2; i++) { + for (j=0; j<6; j++) { + if (sync_required[i][j]) { + rc=tas3001c_sync_biquad(self, i, j); + if (rc < 0) return rc; + } + } + } + if (flags & TAS_BIQUAD_FAST_LOAD) { + tas3001c_fast_load(self,0); + /* now we need to set up the mixers again, + because leaving fast mode resets them. */ + (void)tas3001c_sync_register(self,TAS3001C_REG_BASS); + (void)tas3001c_sync_register(self,TAS3001C_REG_TREBLE); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER1); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER2); + (void)tas3001c_sync_register(self,TAS3001C_REG_VOLUME); + } + } + + return 0; +} + +static int +tas3001c_update_drce( struct tas3001c_data_t *self, + int flags, + struct tas_drce_t *drce) +{ + tas_shadow_t *shadow; + shadow=self->super.shadow; + + shadow[TAS3001C_REG_DRC][1] = 0xc1; + + if (flags & TAS_DRCE_THRESHOLD) { + self->drce_state.threshold=quantize_db(drce->threshold); + shadow[TAS3001C_REG_DRC][2] = db_to_regval(self->drce_state.threshold); + } + + if (flags & TAS_DRCE_ENABLE) { + self->drce_state.enable = drce->enable; + } + + if (!self->drce_state.enable) { + shadow[TAS3001C_REG_DRC][0] = 0xf0; + } + +#ifdef DEBUG_DRCE + printk("DRCE IOCTL: set [ ENABLE:%x THRESH:%x\n", + self->drce_state.enable, + self->drce_state.threshold); + + printk("DRCE IOCTL: reg [ %02x %02x ]\n", + (unsigned char)shadow[TAS3001C_REG_DRC][0], + (unsigned char)shadow[TAS3001C_REG_DRC][1]); +#endif + + return tas3001c_sync_register(self, TAS3001C_REG_DRC); +} + +static int +tas3001c_drce_rw( struct tas3001c_data_t *self, + u_int cmd, + u_long arg) +{ + int rc; + struct tas_drce_ctrl_t drce_ctrl; + void __user *argp = (void __user *)arg; + + if (copy_from_user(&drce_ctrl, argp, sizeof(struct tas_drce_ctrl_t))) + return -EFAULT; + +#ifdef DEBUG_DRCE + printk("DRCE IOCTL: input [ FLAGS:%x ENABLE:%x THRESH:%x\n", + drce_ctrl.flags, + drce_ctrl.data.enable, + drce_ctrl.data.threshold); +#endif + + if (cmd & SIOC_IN) { + rc = tas3001c_update_drce(self, drce_ctrl.flags, &drce_ctrl.data); + if (rc < 0) + return rc; + } + + if (cmd & SIOC_OUT) { + if (drce_ctrl.flags & TAS_DRCE_ENABLE) + drce_ctrl.data.enable = self->drce_state.enable; + + if (drce_ctrl.flags & TAS_DRCE_THRESHOLD) + drce_ctrl.data.threshold = self->drce_state.threshold; + + if (copy_to_user(argp, &drce_ctrl, + sizeof(struct tas_drce_ctrl_t))) { + return -EFAULT; + } + } + + return 0; +} + +static void +tas3001c_update_device_parameters(struct tas3001c_data_t *self) +{ + int i,j; + + if (!self) return; + + if (self->output_id == TAS_OUTPUT_HEADPHONES) { + tas3001c_fast_load(self, 1); + + for (i=0; i<TAS3001C_BIQUAD_CHANNEL_COUNT; i++) { + for (j=0; j<TAS3001C_BIQUAD_FILTER_COUNT; j++) { + tas3001c_write_biquad(self, i, j, &tas3001c_eq_unity); + } + } + + tas3001c_fast_load(self, 0); + + (void)tas3001c_sync_register(self,TAS3001C_REG_BASS); + (void)tas3001c_sync_register(self,TAS3001C_REG_TREBLE); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER1); + (void)tas3001c_sync_register(self,TAS3001C_REG_MIXER2); + (void)tas3001c_sync_register(self,TAS3001C_REG_VOLUME); + + return; + } + + for (i=0; tas3001c_eq_prefs[i]; i++) { + struct tas_eq_pref_t *eq = tas3001c_eq_prefs[i]; + + if (eq->device_id == self->device_id && + (eq->output_id == 0 || eq->output_id == self->output_id) && + (eq->speaker_id == 0 || eq->speaker_id == self->speaker_id)) { + + tas3001c_update_drce(self, TAS_DRCE_ALL, eq->drce); + tas3001c_write_biquad_list(self, eq->filter_count, TAS_BIQUAD_FAST_LOAD, eq->biquads); + + break; + } + } +} + +static void +tas3001c_device_change_handler(void *self) +{ + if (self) + tas3001c_update_device_parameters(self); +} + +static struct work_struct device_change; + +static int +tas3001c_output_device_change( struct tas3001c_data_t *self, + int device_id, + int output_id, + int speaker_id) +{ + self->device_id=device_id; + self->output_id=output_id; + self->speaker_id=speaker_id; + + schedule_work(&device_change); + return 0; +} + +static int +tas3001c_device_ioctl( struct tas3001c_data_t *self, + u_int cmd, + u_long arg) +{ + uint __user *argp = (void __user *)arg; + switch (cmd) { + case TAS_READ_EQ: + case TAS_WRITE_EQ: + return tas3001c_eq_rw(self, cmd, arg); + + case TAS_READ_EQ_LIST: + case TAS_WRITE_EQ_LIST: + return tas3001c_eq_list_rw(self, cmd, arg); + + case TAS_READ_EQ_FILTER_COUNT: + put_user(TAS3001C_BIQUAD_FILTER_COUNT, argp); + return 0; + + case TAS_READ_EQ_CHANNEL_COUNT: + put_user(TAS3001C_BIQUAD_CHANNEL_COUNT, argp); + return 0; + + case TAS_READ_DRCE: + case TAS_WRITE_DRCE: + return tas3001c_drce_rw(self, cmd, arg); + + case TAS_READ_DRCE_CAPS: + put_user(TAS_DRCE_ENABLE | TAS_DRCE_THRESHOLD, argp); + return 0; + + case TAS_READ_DRCE_MIN: + case TAS_READ_DRCE_MAX: { + struct tas_drce_ctrl_t drce_ctrl; + + if (copy_from_user(&drce_ctrl, argp, + sizeof(struct tas_drce_ctrl_t))) { + return -EFAULT; + } + + if (drce_ctrl.flags & TAS_DRCE_THRESHOLD) { + if (cmd == TAS_READ_DRCE_MIN) { + drce_ctrl.data.threshold=-36<<8; + } else { + drce_ctrl.data.threshold=-6<<8; + } + } + + if (copy_to_user(argp, &drce_ctrl, + sizeof(struct tas_drce_ctrl_t))) { + return -EFAULT; + } + } + } + + return -EINVAL; +} + +static int +tas3001c_init_mixer(struct tas3001c_data_t *self) +{ + unsigned char mcr = (1<<6)+(2<<4)+(2<<2); + + /* Make sure something answers on the i2c bus */ + if (tas3001c_write_register(self, TAS3001C_REG_MCR, &mcr, + WRITE_NORMAL|FORCE_WRITE) < 0) + return -1; + + tas3001c_fast_load(self, 1); + + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD0); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD1); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD2); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD3); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD4); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD5); + (void)tas3001c_sync_register(self,TAS3001C_REG_RIGHT_BIQUAD6); + + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD0); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD1); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD2); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD3); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD4); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD5); + (void)tas3001c_sync_register(self,TAS3001C_REG_LEFT_BIQUAD6); + + tas3001c_fast_load(self, 0); + + tas3001c_set_mixer_level(self, SOUND_MIXER_VOLUME, VOL_DEFAULT<<8 | VOL_DEFAULT); + tas3001c_set_mixer_level(self, SOUND_MIXER_PCM, INPUT_DEFAULT<<8 | INPUT_DEFAULT); + tas3001c_set_mixer_level(self, SOUND_MIXER_ALTPCM, 0); + + tas3001c_set_mixer_level(self, SOUND_MIXER_BASS, BASS_DEFAULT); + tas3001c_set_mixer_level(self, SOUND_MIXER_TREBLE, TREBLE_DEFAULT); + + return 0; +} + +static int +tas3001c_uninit_mixer(struct tas3001c_data_t *self) +{ + tas3001c_set_mixer_level(self, SOUND_MIXER_VOLUME, 0); + tas3001c_set_mixer_level(self, SOUND_MIXER_PCM, 0); + tas3001c_set_mixer_level(self, SOUND_MIXER_ALTPCM, 0); + + tas3001c_set_mixer_level(self, SOUND_MIXER_BASS, 0); + tas3001c_set_mixer_level(self, SOUND_MIXER_TREBLE, 0); + + return 0; +} + +static int +tas3001c_init(struct i2c_client *client) +{ + struct tas3001c_data_t *self; + size_t sz = sizeof(*self) + (TAS3001C_REG_MAX*sizeof(tas_shadow_t)); + int i, j; + + self = kmalloc(sz, GFP_KERNEL); + if (!self) + return -ENOMEM; + memset(self, 0, sz); + + self->super.client = client; + self->super.shadow = (tas_shadow_t *)(self+1); + self->output_id = TAS_OUTPUT_HEADPHONES; + + dev_set_drvdata(&client->dev, self); + + for (i = 0; i < TAS3001C_BIQUAD_CHANNEL_COUNT; i++) + for (j = 0; j < TAS3001C_BIQUAD_FILTER_COUNT; j++) + tas3001c_write_biquad_shadow(self, i, j, + &tas3001c_eq_unity); + + INIT_WORK(&device_change, tas3001c_device_change_handler, self); + return 0; +} + +static void +tas3001c_uninit(struct tas3001c_data_t *self) +{ + tas3001c_uninit_mixer(self); + kfree(self); +} + +struct tas_driver_hooks_t tas3001c_hooks = { + .init = (tas_hook_init_t)tas3001c_init, + .post_init = (tas_hook_post_init_t)tas3001c_init_mixer, + .uninit = (tas_hook_uninit_t)tas3001c_uninit, + .get_mixer_level = (tas_hook_get_mixer_level_t)tas3001c_get_mixer_level, + .set_mixer_level = (tas_hook_set_mixer_level_t)tas3001c_set_mixer_level, + .enter_sleep = (tas_hook_enter_sleep_t)tas3001c_enter_sleep, + .leave_sleep = (tas_hook_leave_sleep_t)tas3001c_leave_sleep, + .supported_mixers = (tas_hook_supported_mixers_t)tas3001c_supported_mixers, + .mixer_is_stereo = (tas_hook_mixer_is_stereo_t)tas3001c_mixer_is_stereo, + .stereo_mixers = (tas_hook_stereo_mixers_t)tas3001c_stereo_mixers, + .output_device_change = (tas_hook_output_device_change_t)tas3001c_output_device_change, + .device_ioctl = (tas_hook_device_ioctl_t)tas3001c_device_ioctl +}; diff --git a/sound/oss/dmasound/tas3001c.h b/sound/oss/dmasound/tas3001c.h new file mode 100644 index 0000000..3660da3 --- /dev/null +++ b/sound/oss/dmasound/tas3001c.h @@ -0,0 +1,64 @@ +/* + * Header file for the i2c/i2s based TA3001c sound chip used + * on some Apple hardware. Also known as "tumbler". + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive + * for more details. + * + * Written by Christopher C. Chimelis <chris@debian.org> + */ + +#ifndef _TAS3001C_H_ +#define _TAS3001C_H_ + +#include <linux/types.h> + +#include "tas_common.h" +#include "tas_eq_prefs.h" + +/* + * Macros that correspond to the registers that we write to + * when setting the various values. + */ + +#define TAS3001C_VERSION "0.3" +#define TAS3001C_DATE "20011214" + +#define I2C_DRIVERNAME_TAS3001C "TAS3001c driver V " TAS3001C_VERSION +#define I2C_DRIVERID_TAS3001C (I2C_DRIVERID_TAS_BASE+0) + +extern struct tas_driver_hooks_t tas3001c_hooks; +extern struct tas_gain_t tas3001c_gain; +extern struct tas_eq_pref_t *tas3001c_eq_prefs[]; + +enum tas3001c_reg_t { + TAS3001C_REG_MCR = 0x01, + TAS3001C_REG_DRC = 0x02, + + TAS3001C_REG_VOLUME = 0x04, + TAS3001C_REG_TREBLE = 0x05, + TAS3001C_REG_BASS = 0x06, + TAS3001C_REG_MIXER1 = 0x07, + TAS3001C_REG_MIXER2 = 0x08, + + TAS3001C_REG_LEFT_BIQUAD0 = 0x0a, + TAS3001C_REG_LEFT_BIQUAD1 = 0x0b, + TAS3001C_REG_LEFT_BIQUAD2 = 0x0c, + TAS3001C_REG_LEFT_BIQUAD3 = 0x0d, + TAS3001C_REG_LEFT_BIQUAD4 = 0x0e, + TAS3001C_REG_LEFT_BIQUAD5 = 0x0f, + TAS3001C_REG_LEFT_BIQUAD6 = 0x10, + + TAS3001C_REG_RIGHT_BIQUAD0 = 0x13, + TAS3001C_REG_RIGHT_BIQUAD1 = 0x14, + TAS3001C_REG_RIGHT_BIQUAD2 = 0x15, + TAS3001C_REG_RIGHT_BIQUAD3 = 0x16, + TAS3001C_REG_RIGHT_BIQUAD4 = 0x17, + TAS3001C_REG_RIGHT_BIQUAD5 = 0x18, + TAS3001C_REG_RIGHT_BIQUAD6 = 0x19, + + TAS3001C_REG_MAX = 0x20 +}; + +#endif /* _TAS3001C_H_ */ diff --git a/sound/oss/dmasound/tas3001c_tables.c b/sound/oss/dmasound/tas3001c_tables.c new file mode 100644 index 0000000..1768fa9 --- /dev/null +++ b/sound/oss/dmasound/tas3001c_tables.c @@ -0,0 +1,375 @@ +#include "tas_common.h" +#include "tas_eq_prefs.h" + +static struct tas_drce_t eqp_0e_2_1_drce = { + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -15.33 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_0e_2_1_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0FCAD3, 0xE06A58, 0x0FCAD3, 0xE06B09, 0x0F9657 } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x041731, 0x082E63, 0x041731, 0xFD8D08, 0x02CFBD } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0FFDC7, 0xE0524C, 0x0FBFAA, 0xE0524C, 0x0FBD72 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0F3D35, 0xE228CA, 0x0EC7B2, 0xE228CA, 0x0E04E8 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x0FCEBF, 0xE181C2, 0x0F2656, 0xE181C2, 0x0EF516 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0EC417, 0x073E22, 0x0B0633, 0x073E22, 0x09CA4A } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0FCAD3, 0xE06A58, 0x0FCAD3, 0xE06B09, 0x0F9657 } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x041731, 0x082E63, 0x041731, 0xFD8D08, 0x02CFBD } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0FFDC7, 0xE0524C, 0x0FBFAA, 0xE0524C, 0x0FBD72 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0F3D35, 0xE228CA, 0x0EC7B2, 0xE228CA, 0x0E04E8 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x0FCEBF, 0xE181C2, 0x0F2656, 0xE181C2, 0x0EF516 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0EC417, 0x073E22, 0x0B0633, 0x073E22, 0x09CA4A } } }, +}; + +static struct tas_eq_pref_t eqp_0e_2_1 = { + .sample_rate = 44100, + .device_id = 0x0e, + .output_id = TAS_OUTPUT_EXTERNAL_SPKR, + .speaker_id = 0x01, + + .drce = &eqp_0e_2_1_drce, + + .filter_count = 12, + .biquads = eqp_0e_2_1_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_10_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -12.46 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_10_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0F4A12, 0xE16BDA, 0x0F4A12, 0xE173F0, 0x0E9C3A } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x02DD54, 0x05BAA8, 0x02DD54, 0xF8001D, 0x037532 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0E2FC7, 0xE4D5DC, 0x0D7477, 0xE4D5DC, 0x0BA43F } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0E7899, 0xE67CCA, 0x0D0E93, 0xE67CCA, 0x0B872D } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0F4A12, 0xE16BDA, 0x0F4A12, 0xE173F0, 0x0E9C3A } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x02DD54, 0x05BAA8, 0x02DD54, 0xF8001D, 0x037532 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0E2FC7, 0xE4D5DC, 0x0D7477, 0xE4D5DC, 0x0BA43F } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0E7899, 0xE67CCA, 0x0D0E93, 0xE67CCA, 0x0B872D } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } } }, +}; + +static struct tas_eq_pref_t eqp_10_1_0 = { + .sample_rate = 44100, + .device_id = 0x10, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_10_1_0_drce, + + .filter_count = 12, + .biquads = eqp_10_1_0_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_15_2_1_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -15.33 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_15_2_1_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0FE143, 0xE05204, 0x0FCCC5, 0xE05266, 0x0FAE6B } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x102383, 0xE03A03, 0x0FA325, 0xE03A03, 0x0FC6A8 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0FF2AB, 0xE06285, 0x0FB20A, 0xE06285, 0x0FA4B5 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0F544D, 0xE35971, 0x0D8F3A, 0xE35971, 0x0CE388 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x13E1D3, 0xF3ECB5, 0x042227, 0xF3ECB5, 0x0803FA } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0AC119, 0x034181, 0x078AB1, 0x034181, 0x024BCA } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0FE143, 0xE05204, 0x0FCCC5, 0xE05266, 0x0FAE6B } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x102383, 0xE03A03, 0x0FA325, 0xE03A03, 0x0FC6A8 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0FF2AB, 0xE06285, 0x0FB20A, 0xE06285, 0x0FA4B5 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0F544D, 0xE35971, 0x0D8F3A, 0xE35971, 0x0CE388 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x13E1D3, 0xF3ECB5, 0x042227, 0xF3ECB5, 0x0803FA } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0AC119, 0x034181, 0x078AB1, 0x034181, 0x024BCA } } }, +}; + +static struct tas_eq_pref_t eqp_15_2_1 = { + .sample_rate = 44100, + .device_id = 0x15, + .output_id = TAS_OUTPUT_EXTERNAL_SPKR, + .speaker_id = 0x01, + + .drce = &eqp_15_2_1_drce, + + .filter_count = 12, + .biquads = eqp_15_2_1_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_15_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = 0.0 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_15_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0FAD08, 0xE0A5EF, 0x0FAD08, 0xE0A79D, 0x0F5BBE } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x04B38D, 0x09671B, 0x04B38D, 0x000F71, 0x02BEC5 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0FDD32, 0xE0A56F, 0x0F8A69, 0xE0A56F, 0x0F679C } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0FD284, 0xE135FB, 0x0F2161, 0xE135FB, 0x0EF3E5 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x0E81B1, 0xE6283F, 0x0CE49D, 0xE6283F, 0x0B664F } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0F2D62, 0xE98797, 0x0D1E19, 0xE98797, 0x0C4B7B } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0FAD08, 0xE0A5EF, 0x0FAD08, 0xE0A79D, 0x0F5BBE } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x04B38D, 0x09671B, 0x04B38D, 0x000F71, 0x02BEC5 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0FDD32, 0xE0A56F, 0x0F8A69, 0xE0A56F, 0x0F679C } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0FD284, 0xE135FB, 0x0F2161, 0xE135FB, 0x0EF3E5 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x0E81B1, 0xE6283F, 0x0CE49D, 0xE6283F, 0x0B664F } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0F2D62, 0xE98797, 0x0D1E19, 0xE98797, 0x0C4B7B } } }, +}; + +static struct tas_eq_pref_t eqp_15_1_0 = { + .sample_rate = 44100, + .device_id = 0x15, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_15_1_0_drce, + + .filter_count = 12, + .biquads = eqp_15_1_0_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_0f_2_1_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -15.33 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_0f_2_1_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0FE143, 0xE05204, 0x0FCCC5, 0xE05266, 0x0FAE6B } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x102383, 0xE03A03, 0x0FA325, 0xE03A03, 0x0FC6A8 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0FF2AB, 0xE06285, 0x0FB20A, 0xE06285, 0x0FA4B5 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0F544D, 0xE35971, 0x0D8F3A, 0xE35971, 0x0CE388 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x13E1D3, 0xF3ECB5, 0x042227, 0xF3ECB5, 0x0803FA } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0AC119, 0x034181, 0x078AB1, 0x034181, 0x024BCA } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0FE143, 0xE05204, 0x0FCCC5, 0xE05266, 0x0FAE6B } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x102383, 0xE03A03, 0x0FA325, 0xE03A03, 0x0FC6A8 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0FF2AB, 0xE06285, 0x0FB20A, 0xE06285, 0x0FA4B5 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0F544D, 0xE35971, 0x0D8F3A, 0xE35971, 0x0CE388 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x13E1D3, 0xF3ECB5, 0x042227, 0xF3ECB5, 0x0803FA } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0AC119, 0x034181, 0x078AB1, 0x034181, 0x024BCA } } }, +}; + +static struct tas_eq_pref_t eqp_0f_2_1 = { + .sample_rate = 44100, + .device_id = 0x0f, + .output_id = TAS_OUTPUT_EXTERNAL_SPKR, + .speaker_id = 0x01, + + .drce = &eqp_0f_2_1_drce, + + .filter_count = 12, + .biquads = eqp_0f_2_1_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_0f_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -15.33 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_0f_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0FCAD3, 0xE06A58, 0x0FCAD3, 0xE06B09, 0x0F9657 } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x041731, 0x082E63, 0x041731, 0xFD8D08, 0x02CFBD } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0FFDC7, 0xE0524C, 0x0FBFAA, 0xE0524C, 0x0FBD72 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0F3D35, 0xE228CA, 0x0EC7B2, 0xE228CA, 0x0E04E8 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x0FCEBF, 0xE181C2, 0x0F2656, 0xE181C2, 0x0EF516 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0EC417, 0x073E22, 0x0B0633, 0x073E22, 0x09CA4A } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0FCAD3, 0xE06A58, 0x0FCAD3, 0xE06B09, 0x0F9657 } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x041731, 0x082E63, 0x041731, 0xFD8D08, 0x02CFBD } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0FFDC7, 0xE0524C, 0x0FBFAA, 0xE0524C, 0x0FBD72 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0F3D35, 0xE228CA, 0x0EC7B2, 0xE228CA, 0x0E04E8 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x0FCEBF, 0xE181C2, 0x0F2656, 0xE181C2, 0x0EF516 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0EC417, 0x073E22, 0x0B0633, 0x073E22, 0x09CA4A } } }, +}; + +static struct tas_eq_pref_t eqp_0f_1_0 = { + .sample_rate = 44100, + .device_id = 0x0f, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_0f_1_0_drce, + + .filter_count = 12, + .biquads = eqp_0f_1_0_biquads +}; + +/* ======================================================================== */ + +static uint tas3001c_master_tab[]={ + 0x0, 0x75, 0x9c, 0xbb, + 0xdb, 0xfb, 0x11e, 0x143, + 0x16b, 0x196, 0x1c3, 0x1f5, + 0x229, 0x263, 0x29f, 0x2e1, + 0x328, 0x373, 0x3c5, 0x41b, + 0x478, 0x4dc, 0x547, 0x5b8, + 0x633, 0x6b5, 0x740, 0x7d5, + 0x873, 0x91c, 0x9d2, 0xa92, + 0xb5e, 0xc39, 0xd22, 0xe19, + 0xf20, 0x1037, 0x1161, 0x129e, + 0x13ed, 0x1551, 0x16ca, 0x185d, + 0x1a08, 0x1bcc, 0x1dac, 0x1fa7, + 0x21c1, 0x23fa, 0x2655, 0x28d6, + 0x2b7c, 0x2e4a, 0x3141, 0x3464, + 0x37b4, 0x3b35, 0x3ee9, 0x42d3, + 0x46f6, 0x4b53, 0x4ff0, 0x54ce, + 0x59f2, 0x5f5f, 0x6519, 0x6b24, + 0x7183, 0x783c, 0x7f53, 0x86cc, + 0x8ead, 0x96fa, 0x9fba, 0xa8f2, + 0xb2a7, 0xbce1, 0xc7a5, 0xd2fa, + 0xdee8, 0xeb75, 0xf8aa, 0x1068e, + 0x1152a, 0x12487, 0x134ad, 0x145a5, + 0x1577b, 0x16a37, 0x17df5, 0x192bd, + 0x1a890, 0x1bf7b, 0x1d78d, 0x1f0d1, + 0x20b55, 0x22727, 0x24456, 0x262f2, + 0x2830b +}; + +static uint tas3001c_mixer_tab[]={ + 0x0, 0x748, 0x9be, 0xbaf, + 0xda4, 0xfb1, 0x11de, 0x1431, + 0x16ad, 0x1959, 0x1c37, 0x1f4b, + 0x2298, 0x2628, 0x29fb, 0x2e12, + 0x327d, 0x3734, 0x3c47, 0x41b4, + 0x4787, 0x4dbe, 0x546d, 0x5b86, + 0x632e, 0x6b52, 0x7400, 0x7d54, + 0x873b, 0x91c6, 0x9d1a, 0xa920, + 0xb5e5, 0xc38c, 0xd21b, 0xe18f, + 0xf1f5, 0x1036a, 0x1160f, 0x129d6, + 0x13ed0, 0x1550c, 0x16ca0, 0x185c9, + 0x1a07b, 0x1bcc3, 0x1dab9, 0x1fa75, + 0x21c0f, 0x23fa3, 0x26552, 0x28d64, + 0x2b7c9, 0x2e4a2, 0x31411, 0x3463b, + 0x37b44, 0x3b353, 0x3ee94, 0x42d30, + 0x46f55, 0x4b533, 0x4fefc, 0x54ce5, + 0x59f25, 0x5f5f6, 0x65193, 0x6b23c, + 0x71835, 0x783c3, 0x7f52c, 0x86cc0, + 0x8eacc, 0x96fa5, 0x9fba0, 0xa8f1a, + 0xb2a71, 0xbce0a, 0xc7a4a, 0xd2fa0, + 0xdee7b, 0xeb752, 0xf8a9f, 0x1068e4, + 0x1152a3, 0x12486a, 0x134ac8, 0x145a55, + 0x1577ac, 0x16a370, 0x17df51, 0x192bc2, + 0x1a88f8, 0x1bf7b7, 0x1d78c9, 0x1f0d04, + 0x20b542, 0x227268, 0x244564, 0x262f26, + 0x2830af +}; + +static uint tas3001c_treble_tab[]={ + 0x96, 0x95, 0x95, 0x94, + 0x93, 0x92, 0x92, 0x91, + 0x90, 0x90, 0x8f, 0x8e, + 0x8d, 0x8d, 0x8c, 0x8b, + 0x8a, 0x8a, 0x89, 0x88, + 0x88, 0x87, 0x86, 0x85, + 0x85, 0x84, 0x83, 0x83, + 0x82, 0x81, 0x80, 0x80, + 0x7f, 0x7e, 0x7e, 0x7d, + 0x7c, 0x7b, 0x7b, 0x7a, + 0x79, 0x78, 0x78, 0x77, + 0x76, 0x76, 0x75, 0x74, + 0x73, 0x73, 0x72, 0x71, + 0x71, 0x70, 0x6e, 0x6d, + 0x6d, 0x6c, 0x6b, 0x6a, + 0x69, 0x68, 0x67, 0x66, + 0x65, 0x63, 0x62, 0x62, + 0x60, 0x5f, 0x5d, 0x5c, + 0x5a, 0x58, 0x56, 0x55, + 0x53, 0x51, 0x4f, 0x4c, + 0x4a, 0x48, 0x45, 0x43, + 0x40, 0x3d, 0x3a, 0x37, + 0x35, 0x32, 0x2e, 0x2a, + 0x27, 0x22, 0x1e, 0x1a, + 0x15, 0x11, 0xc, 0x7, + 0x1 +}; + +static uint tas3001c_bass_tab[]={ + 0x86, 0x83, 0x81, 0x7f, + 0x7d, 0x7b, 0x79, 0x78, + 0x76, 0x75, 0x74, 0x72, + 0x71, 0x6f, 0x6e, 0x6d, + 0x6c, 0x6b, 0x69, 0x67, + 0x65, 0x64, 0x61, 0x60, + 0x5e, 0x5d, 0x5c, 0x5b, + 0x5a, 0x59, 0x58, 0x57, + 0x56, 0x55, 0x55, 0x54, + 0x53, 0x52, 0x50, 0x4f, + 0x4d, 0x4c, 0x4b, 0x49, + 0x47, 0x45, 0x44, 0x42, + 0x41, 0x3f, 0x3e, 0x3d, + 0x3c, 0x3b, 0x39, 0x38, + 0x37, 0x36, 0x35, 0x34, + 0x33, 0x31, 0x30, 0x2f, + 0x2e, 0x2c, 0x2b, 0x2b, + 0x29, 0x28, 0x27, 0x26, + 0x25, 0x24, 0x22, 0x21, + 0x20, 0x1e, 0x1c, 0x19, + 0x18, 0x18, 0x17, 0x16, + 0x15, 0x14, 0x13, 0x12, + 0x11, 0x10, 0xf, 0xe, + 0xd, 0xb, 0xa, 0x9, + 0x8, 0x6, 0x4, 0x2, + 0x1 +}; + +struct tas_gain_t tas3001c_gain = { + .master = tas3001c_master_tab, + .treble = tas3001c_treble_tab, + .bass = tas3001c_bass_tab, + .mixer = tas3001c_mixer_tab +}; + +struct tas_eq_pref_t *tas3001c_eq_prefs[]={ + &eqp_0e_2_1, + &eqp_10_1_0, + &eqp_15_2_1, + &eqp_15_1_0, + &eqp_0f_2_1, + &eqp_0f_1_0, + NULL +}; diff --git a/sound/oss/dmasound/tas3004.c b/sound/oss/dmasound/tas3004.c new file mode 100644 index 0000000..82eaaca --- /dev/null +++ b/sound/oss/dmasound/tas3004.c @@ -0,0 +1,1140 @@ +/* + * Driver for the i2c/i2s based TA3004 sound chip used + * on some Apple hardware. Also known as "snapper". + * + * Tobias Sargeant <tobias.sargeant@bigpond.com> + * Based upon tas3001c.c by Christopher C. Chimelis <chris@debian.org>: + * + * Input support by Renzo Davoli <renzo@cs.unibo.it> + * + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/proc_fs.h> +#include <linux/ioport.h> +#include <linux/sysctl.h> +#include <linux/types.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> + +#include <asm/uaccess.h> +#include <asm/errno.h> +#include <asm/io.h> +#include <asm/prom.h> + +#include "dmasound.h" +#include "tas_common.h" +#include "tas3004.h" + +#include "tas_ioctl.h" + +/* #define DEBUG_DRCE */ + +#define TAS3004_BIQUAD_FILTER_COUNT 7 +#define TAS3004_BIQUAD_CHANNEL_COUNT 2 + +#define VOL_DEFAULT (100 * 4 / 5) +#define INPUT_DEFAULT (100 * 4 / 5) +#define BASS_DEFAULT (100 / 2) +#define TREBLE_DEFAULT (100 / 2) + +struct tas3004_data_t { + struct tas_data_t super; + int device_id; + int output_id; + int speaker_id; + struct tas_drce_t drce_state; +}; + +#define MAKE_TIME(sec,usec) (((sec)<<12) + (50000+(usec/10)*(1<<12))/100000) + +#define MAKE_RATIO(i,f) (((i)<<8) + ((500+(f)*(1<<8))/1000)) + + +static const union tas_biquad_t tas3004_eq_unity = { + .buf = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 }, +}; + + +static const struct tas_drce_t tas3004_drce_min = { + .enable = 1, + .above = { .val = MAKE_RATIO(16,0), .expand = 0 }, + .below = { .val = MAKE_RATIO(2,0), .expand = 0 }, + .threshold = -0x59a0, + .energy = MAKE_TIME(0, 1700), + .attack = MAKE_TIME(0, 1700), + .decay = MAKE_TIME(0, 1700), +}; + + +static const struct tas_drce_t tas3004_drce_max = { + .enable = 1, + .above = { .val = MAKE_RATIO(1,500), .expand = 1 }, + .below = { .val = MAKE_RATIO(2,0), .expand = 1 }, + .threshold = -0x0, + .energy = MAKE_TIME(2,400000), + .attack = MAKE_TIME(2,400000), + .decay = MAKE_TIME(2,400000), +}; + + +static const unsigned short time_constants[]={ + MAKE_TIME(0, 1700), + MAKE_TIME(0, 3500), + MAKE_TIME(0, 6700), + MAKE_TIME(0, 13000), + MAKE_TIME(0, 26000), + MAKE_TIME(0, 53000), + MAKE_TIME(0,106000), + MAKE_TIME(0,212000), + MAKE_TIME(0,425000), + MAKE_TIME(0,850000), + MAKE_TIME(1,700000), + MAKE_TIME(2,400000), +}; + +static const unsigned short above_threshold_compression_ratio[]={ + MAKE_RATIO( 1, 70), + MAKE_RATIO( 1,140), + MAKE_RATIO( 1,230), + MAKE_RATIO( 1,330), + MAKE_RATIO( 1,450), + MAKE_RATIO( 1,600), + MAKE_RATIO( 1,780), + MAKE_RATIO( 2, 0), + MAKE_RATIO( 2,290), + MAKE_RATIO( 2,670), + MAKE_RATIO( 3,200), + MAKE_RATIO( 4, 0), + MAKE_RATIO( 5,330), + MAKE_RATIO( 8, 0), + MAKE_RATIO(16, 0), +}; + +static const unsigned short above_threshold_expansion_ratio[]={ + MAKE_RATIO(1, 60), + MAKE_RATIO(1,130), + MAKE_RATIO(1,190), + MAKE_RATIO(1,250), + MAKE_RATIO(1,310), + MAKE_RATIO(1,380), + MAKE_RATIO(1,440), + MAKE_RATIO(1,500) +}; + +static const unsigned short below_threshold_compression_ratio[]={ + MAKE_RATIO(1, 70), + MAKE_RATIO(1,140), + MAKE_RATIO(1,230), + MAKE_RATIO(1,330), + MAKE_RATIO(1,450), + MAKE_RATIO(1,600), + MAKE_RATIO(1,780), + MAKE_RATIO(2, 0) +}; + +static const unsigned short below_threshold_expansion_ratio[]={ + MAKE_RATIO(1, 60), + MAKE_RATIO(1,130), + MAKE_RATIO(1,190), + MAKE_RATIO(1,250), + MAKE_RATIO(1,310), + MAKE_RATIO(1,380), + MAKE_RATIO(1,440), + MAKE_RATIO(1,500), + MAKE_RATIO(1,560), + MAKE_RATIO(1,630), + MAKE_RATIO(1,690), + MAKE_RATIO(1,750), + MAKE_RATIO(1,810), + MAKE_RATIO(1,880), + MAKE_RATIO(1,940), + MAKE_RATIO(2, 0) +}; + +static inline int +search( unsigned short val, + const unsigned short *arr, + const int arrsize) { + /* + * This could be a binary search, but for small tables, + * a linear search is likely to be faster + */ + + int i; + + for (i=0; i < arrsize; i++) + if (arr[i] >= val) + goto _1; + return arrsize-1; + _1: + if (i == 0) + return 0; + return (arr[i]-val < val-arr[i-1]) ? i : i-1; +} + +#define SEARCH(a, b) search(a, b, ARRAY_SIZE(b)) + +static inline int +time_index(unsigned short time) +{ + return SEARCH(time, time_constants); +} + + +static inline int +above_threshold_compression_index(unsigned short ratio) +{ + return SEARCH(ratio, above_threshold_compression_ratio); +} + + +static inline int +above_threshold_expansion_index(unsigned short ratio) +{ + return SEARCH(ratio, above_threshold_expansion_ratio); +} + + +static inline int +below_threshold_compression_index(unsigned short ratio) +{ + return SEARCH(ratio, below_threshold_compression_ratio); +} + + +static inline int +below_threshold_expansion_index(unsigned short ratio) +{ + return SEARCH(ratio, below_threshold_expansion_ratio); +} + +static inline unsigned char db_to_regval(short db) { + int r=0; + + r=(db+0x59a0) / 0x60; + + if (r < 0x91) return 0x91; + if (r > 0xef) return 0xef; + return r; +} + +static inline short quantize_db(short db) +{ + return db_to_regval(db) * 0x60 - 0x59a0; +} + +static inline int +register_width(enum tas3004_reg_t r) +{ + switch(r) { + case TAS3004_REG_MCR: + case TAS3004_REG_TREBLE: + case TAS3004_REG_BASS: + case TAS3004_REG_ANALOG_CTRL: + case TAS3004_REG_TEST1: + case TAS3004_REG_TEST2: + case TAS3004_REG_MCR2: + return 1; + + case TAS3004_REG_LEFT_LOUD_BIQUAD_GAIN: + case TAS3004_REG_RIGHT_LOUD_BIQUAD_GAIN: + return 3; + + case TAS3004_REG_DRC: + case TAS3004_REG_VOLUME: + return 6; + + case TAS3004_REG_LEFT_MIXER: + case TAS3004_REG_RIGHT_MIXER: + return 9; + + case TAS3004_REG_TEST: + return 10; + + case TAS3004_REG_LEFT_BIQUAD0: + case TAS3004_REG_LEFT_BIQUAD1: + case TAS3004_REG_LEFT_BIQUAD2: + case TAS3004_REG_LEFT_BIQUAD3: + case TAS3004_REG_LEFT_BIQUAD4: + case TAS3004_REG_LEFT_BIQUAD5: + case TAS3004_REG_LEFT_BIQUAD6: + + case TAS3004_REG_RIGHT_BIQUAD0: + case TAS3004_REG_RIGHT_BIQUAD1: + case TAS3004_REG_RIGHT_BIQUAD2: + case TAS3004_REG_RIGHT_BIQUAD3: + case TAS3004_REG_RIGHT_BIQUAD4: + case TAS3004_REG_RIGHT_BIQUAD5: + case TAS3004_REG_RIGHT_BIQUAD6: + + case TAS3004_REG_LEFT_LOUD_BIQUAD: + case TAS3004_REG_RIGHT_LOUD_BIQUAD: + return 15; + + default: + return 0; + } +} + +static int +tas3004_write_register( struct tas3004_data_t *self, + enum tas3004_reg_t reg_num, + char *data, + uint write_mode) +{ + if (reg_num==TAS3004_REG_MCR || + reg_num==TAS3004_REG_BASS || + reg_num==TAS3004_REG_TREBLE || + reg_num==TAS3004_REG_ANALOG_CTRL) { + return tas_write_byte_register(&self->super, + (uint)reg_num, + *data, + write_mode); + } else { + return tas_write_register(&self->super, + (uint)reg_num, + register_width(reg_num), + data, + write_mode); + } +} + +static int +tas3004_sync_register( struct tas3004_data_t *self, + enum tas3004_reg_t reg_num) +{ + if (reg_num==TAS3004_REG_MCR || + reg_num==TAS3004_REG_BASS || + reg_num==TAS3004_REG_TREBLE || + reg_num==TAS3004_REG_ANALOG_CTRL) { + return tas_sync_byte_register(&self->super, + (uint)reg_num, + register_width(reg_num)); + } else { + return tas_sync_register(&self->super, + (uint)reg_num, + register_width(reg_num)); + } +} + +static int +tas3004_read_register( struct tas3004_data_t *self, + enum tas3004_reg_t reg_num, + char *data, + uint write_mode) +{ + return tas_read_register(&self->super, + (uint)reg_num, + register_width(reg_num), + data); +} + +static inline int +tas3004_fast_load(struct tas3004_data_t *self, int fast) +{ + if (fast) + self->super.shadow[TAS3004_REG_MCR][0] |= 0x80; + else + self->super.shadow[TAS3004_REG_MCR][0] &= 0x7f; + return tas3004_sync_register(self,TAS3004_REG_MCR); +} + +static uint +tas3004_supported_mixers(struct tas3004_data_t *self) +{ + return SOUND_MASK_VOLUME | + SOUND_MASK_PCM | + SOUND_MASK_ALTPCM | + SOUND_MASK_IMIX | + SOUND_MASK_TREBLE | + SOUND_MASK_BASS | + SOUND_MASK_MIC | + SOUND_MASK_LINE; +} + +static int +tas3004_mixer_is_stereo(struct tas3004_data_t *self, int mixer) +{ + switch(mixer) { + case SOUND_MIXER_VOLUME: + case SOUND_MIXER_PCM: + case SOUND_MIXER_ALTPCM: + case SOUND_MIXER_IMIX: + return 1; + default: + return 0; + } +} + +static uint +tas3004_stereo_mixers(struct tas3004_data_t *self) +{ + uint r = tas3004_supported_mixers(self); + uint i; + + for (i=1; i<SOUND_MIXER_NRDEVICES; i++) + if (r&(1<<i) && !tas3004_mixer_is_stereo(self,i)) + r &= ~(1<<i); + return r; +} + +static int +tas3004_get_mixer_level(struct tas3004_data_t *self, int mixer, uint *level) +{ + if (!self) + return -1; + + *level = self->super.mixer[mixer]; + + return 0; +} + +static int +tas3004_set_mixer_level(struct tas3004_data_t *self, int mixer, uint level) +{ + int rc; + tas_shadow_t *shadow; + uint temp; + uint offset=0; + + if (!self) + return -1; + + shadow = self->super.shadow; + + if (!tas3004_mixer_is_stereo(self,mixer)) + level = tas_mono_to_stereo(level); + switch(mixer) { + case SOUND_MIXER_VOLUME: + temp = tas3004_gain.master[level&0xff]; + SET_4_20(shadow[TAS3004_REG_VOLUME], 0, temp); + temp = tas3004_gain.master[(level>>8)&0xff]; + SET_4_20(shadow[TAS3004_REG_VOLUME], 3, temp); + rc = tas3004_sync_register(self,TAS3004_REG_VOLUME); + break; + case SOUND_MIXER_IMIX: + offset += 3; + case SOUND_MIXER_ALTPCM: + offset += 3; + case SOUND_MIXER_PCM: + /* + * Don't load these in fast mode. The documentation + * says it can be done in either mode, but testing it + * shows that fast mode produces ugly clicking. + */ + /* tas3004_fast_load(self,1); */ + temp = tas3004_gain.mixer[level&0xff]; + SET_4_20(shadow[TAS3004_REG_LEFT_MIXER], offset, temp); + temp = tas3004_gain.mixer[(level>>8)&0xff]; + SET_4_20(shadow[TAS3004_REG_RIGHT_MIXER], offset, temp); + rc = tas3004_sync_register(self,TAS3004_REG_LEFT_MIXER); + if (rc == 0) + rc=tas3004_sync_register(self,TAS3004_REG_RIGHT_MIXER); + /* tas3004_fast_load(self,0); */ + break; + case SOUND_MIXER_TREBLE: + temp = tas3004_gain.treble[level&0xff]; + shadow[TAS3004_REG_TREBLE][0]=temp&0xff; + rc = tas3004_sync_register(self,TAS3004_REG_TREBLE); + break; + case SOUND_MIXER_BASS: + temp = tas3004_gain.bass[level&0xff]; + shadow[TAS3004_REG_BASS][0]=temp&0xff; + rc = tas3004_sync_register(self,TAS3004_REG_BASS); + break; + case SOUND_MIXER_MIC: + if ((level&0xff)>0) { + software_input_volume = SW_INPUT_VOLUME_SCALE * (level&0xff); + if (self->super.mixer[mixer] == 0) { + self->super.mixer[SOUND_MIXER_LINE] = 0; + shadow[TAS3004_REG_ANALOG_CTRL][0]=0xc2; + rc = tas3004_sync_register(self,TAS3004_REG_ANALOG_CTRL); + } else rc=0; + } else { + self->super.mixer[SOUND_MIXER_LINE] = SW_INPUT_VOLUME_DEFAULT; + software_input_volume = SW_INPUT_VOLUME_SCALE * + (self->super.mixer[SOUND_MIXER_LINE]&0xff); + shadow[TAS3004_REG_ANALOG_CTRL][0]=0x00; + rc = tas3004_sync_register(self,TAS3004_REG_ANALOG_CTRL); + } + break; + case SOUND_MIXER_LINE: + if (self->super.mixer[SOUND_MIXER_MIC] == 0) { + software_input_volume = SW_INPUT_VOLUME_SCALE * (level&0xff); + rc=0; + } + break; + default: + rc = -1; + break; + } + if (rc < 0) + return rc; + self->super.mixer[mixer] = level; + + return 0; +} + +static int +tas3004_leave_sleep(struct tas3004_data_t *self) +{ + unsigned char mcr = (1<<6)+(2<<4)+(2<<2); + + if (!self) + return -1; + + /* Make sure something answers on the i2c bus */ + if (tas3004_write_register(self, TAS3004_REG_MCR, &mcr, + WRITE_NORMAL | FORCE_WRITE) < 0) + return -1; + + tas3004_fast_load(self, 1); + + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD0); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD1); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD2); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD3); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD4); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD5); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD6); + + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD0); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD1); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD2); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD3); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD4); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD5); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD6); + + tas3004_fast_load(self, 0); + + (void)tas3004_sync_register(self,TAS3004_REG_VOLUME); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_MIXER); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_MIXER); + (void)tas3004_sync_register(self,TAS3004_REG_TREBLE); + (void)tas3004_sync_register(self,TAS3004_REG_BASS); + (void)tas3004_sync_register(self,TAS3004_REG_ANALOG_CTRL); + + return 0; +} + +static int +tas3004_enter_sleep(struct tas3004_data_t *self) +{ + if (!self) + return -1; + return 0; +} + +static int +tas3004_sync_biquad( struct tas3004_data_t *self, + u_int channel, + u_int filter) +{ + enum tas3004_reg_t reg; + + if (channel >= TAS3004_BIQUAD_CHANNEL_COUNT || + filter >= TAS3004_BIQUAD_FILTER_COUNT) return -EINVAL; + + reg=( channel ? TAS3004_REG_RIGHT_BIQUAD0 : TAS3004_REG_LEFT_BIQUAD0 ) + filter; + + return tas3004_sync_register(self,reg); +} + +static int +tas3004_write_biquad_shadow( struct tas3004_data_t *self, + u_int channel, + u_int filter, + const union tas_biquad_t *biquad) +{ + tas_shadow_t *shadow=self->super.shadow; + enum tas3004_reg_t reg; + + if (channel >= TAS3004_BIQUAD_CHANNEL_COUNT || + filter >= TAS3004_BIQUAD_FILTER_COUNT) return -EINVAL; + + reg=( channel ? TAS3004_REG_RIGHT_BIQUAD0 : TAS3004_REG_LEFT_BIQUAD0 ) + filter; + + SET_4_20(shadow[reg], 0,biquad->coeff.b0); + SET_4_20(shadow[reg], 3,biquad->coeff.b1); + SET_4_20(shadow[reg], 6,biquad->coeff.b2); + SET_4_20(shadow[reg], 9,biquad->coeff.a1); + SET_4_20(shadow[reg],12,biquad->coeff.a2); + + return 0; +} + +static int +tas3004_write_biquad( struct tas3004_data_t *self, + u_int channel, + u_int filter, + const union tas_biquad_t *biquad) +{ + int rc; + + rc=tas3004_write_biquad_shadow(self, channel, filter, biquad); + if (rc < 0) return rc; + + return tas3004_sync_biquad(self, channel, filter); +} + +static int +tas3004_write_biquad_list( struct tas3004_data_t *self, + u_int filter_count, + u_int flags, + struct tas_biquad_ctrl_t *biquads) +{ + int i; + int rc; + + if (flags & TAS_BIQUAD_FAST_LOAD) tas3004_fast_load(self,1); + + for (i=0; i<filter_count; i++) { + rc=tas3004_write_biquad(self, + biquads[i].channel, + biquads[i].filter, + &biquads[i].data); + if (rc < 0) break; + } + + if (flags & TAS_BIQUAD_FAST_LOAD) tas3004_fast_load(self,0); + + return rc; +} + +static int +tas3004_read_biquad( struct tas3004_data_t *self, + u_int channel, + u_int filter, + union tas_biquad_t *biquad) +{ + tas_shadow_t *shadow=self->super.shadow; + enum tas3004_reg_t reg; + + if (channel >= TAS3004_BIQUAD_CHANNEL_COUNT || + filter >= TAS3004_BIQUAD_FILTER_COUNT) return -EINVAL; + + reg=( channel ? TAS3004_REG_RIGHT_BIQUAD0 : TAS3004_REG_LEFT_BIQUAD0 ) + filter; + + biquad->coeff.b0=GET_4_20(shadow[reg], 0); + biquad->coeff.b1=GET_4_20(shadow[reg], 3); + biquad->coeff.b2=GET_4_20(shadow[reg], 6); + biquad->coeff.a1=GET_4_20(shadow[reg], 9); + biquad->coeff.a2=GET_4_20(shadow[reg],12); + + return 0; +} + +static int +tas3004_eq_rw( struct tas3004_data_t *self, + u_int cmd, + u_long arg) +{ + void __user *argp = (void __user *)arg; + int rc; + struct tas_biquad_ctrl_t biquad; + + if (copy_from_user((void *)&biquad, argp, sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + + if (cmd & SIOC_IN) { + rc=tas3004_write_biquad(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + } + + if (cmd & SIOC_OUT) { + rc=tas3004_read_biquad(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + + if (copy_to_user(argp, &biquad, sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + + } + return 0; +} + +static int +tas3004_eq_list_rw( struct tas3004_data_t *self, + u_int cmd, + u_long arg) +{ + int rc = 0; + int filter_count; + int flags; + int i,j; + char sync_required[TAS3004_BIQUAD_CHANNEL_COUNT][TAS3004_BIQUAD_FILTER_COUNT]; + struct tas_biquad_ctrl_t biquad; + struct tas_biquad_ctrl_list_t __user *argp = (void __user *)arg; + + memset(sync_required,0,sizeof(sync_required)); + + if (copy_from_user(&filter_count, &argp->filter_count, sizeof(int))) + return -EFAULT; + + if (copy_from_user(&flags, &argp->flags, sizeof(int))) + return -EFAULT; + + if (cmd & SIOC_IN) { + } + + for (i=0; i < filter_count; i++) { + if (copy_from_user(&biquad, &argp->biquads[i], + sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + + if (cmd & SIOC_IN) { + sync_required[biquad.channel][biquad.filter]=1; + rc=tas3004_write_biquad_shadow(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + } + + if (cmd & SIOC_OUT) { + rc=tas3004_read_biquad(self, biquad.channel, biquad.filter, &biquad.data); + if (rc != 0) return rc; + + if (copy_to_user(&argp->biquads[i], &biquad, + sizeof(struct tas_biquad_ctrl_t))) { + return -EFAULT; + } + } + } + + if (cmd & SIOC_IN) { + /* + * This is OK for the tas3004. For the + * tas3001c, going into fast load mode causes + * the treble and bass to be reset to 0dB, and + * volume controls to be muted. + */ + if (flags & TAS_BIQUAD_FAST_LOAD) tas3004_fast_load(self,1); + for (i=0; i<TAS3004_BIQUAD_CHANNEL_COUNT; i++) { + for (j=0; j<TAS3004_BIQUAD_FILTER_COUNT; j++) { + if (sync_required[i][j]) { + rc=tas3004_sync_biquad(self, i, j); + if (rc < 0) goto out; + } + } + } + out: + if (flags & TAS_BIQUAD_FAST_LOAD) + tas3004_fast_load(self,0); + } + + return rc; +} + +static int +tas3004_update_drce( struct tas3004_data_t *self, + int flags, + struct tas_drce_t *drce) +{ + tas_shadow_t *shadow; + int i; + shadow=self->super.shadow; + + if (flags & TAS_DRCE_ABOVE_RATIO) { + self->drce_state.above.expand = drce->above.expand; + if (drce->above.val == (1<<8)) { + self->drce_state.above.val = 1<<8; + shadow[TAS3004_REG_DRC][0] = 0x02; + + } else if (drce->above.expand) { + i=above_threshold_expansion_index(drce->above.val); + self->drce_state.above.val=above_threshold_expansion_ratio[i]; + shadow[TAS3004_REG_DRC][0] = 0x0a + (i<<3); + } else { + i=above_threshold_compression_index(drce->above.val); + self->drce_state.above.val=above_threshold_compression_ratio[i]; + shadow[TAS3004_REG_DRC][0] = 0x08 + (i<<3); + } + } + + if (flags & TAS_DRCE_BELOW_RATIO) { + self->drce_state.below.expand = drce->below.expand; + if (drce->below.val == (1<<8)) { + self->drce_state.below.val = 1<<8; + shadow[TAS3004_REG_DRC][1] = 0x02; + + } else if (drce->below.expand) { + i=below_threshold_expansion_index(drce->below.val); + self->drce_state.below.val=below_threshold_expansion_ratio[i]; + shadow[TAS3004_REG_DRC][1] = 0x08 + (i<<3); + } else { + i=below_threshold_compression_index(drce->below.val); + self->drce_state.below.val=below_threshold_compression_ratio[i]; + shadow[TAS3004_REG_DRC][1] = 0x0a + (i<<3); + } + } + + if (flags & TAS_DRCE_THRESHOLD) { + self->drce_state.threshold=quantize_db(drce->threshold); + shadow[TAS3004_REG_DRC][2] = db_to_regval(self->drce_state.threshold); + } + + if (flags & TAS_DRCE_ENERGY) { + i=time_index(drce->energy); + self->drce_state.energy=time_constants[i]; + shadow[TAS3004_REG_DRC][3] = 0x40 + (i<<4); + } + + if (flags & TAS_DRCE_ATTACK) { + i=time_index(drce->attack); + self->drce_state.attack=time_constants[i]; + shadow[TAS3004_REG_DRC][4] = 0x40 + (i<<4); + } + + if (flags & TAS_DRCE_DECAY) { + i=time_index(drce->decay); + self->drce_state.decay=time_constants[i]; + shadow[TAS3004_REG_DRC][5] = 0x40 + (i<<4); + } + + if (flags & TAS_DRCE_ENABLE) { + self->drce_state.enable = drce->enable; + } + + if (!self->drce_state.enable) { + shadow[TAS3004_REG_DRC][0] |= 0x01; + } + +#ifdef DEBUG_DRCE + printk("DRCE: set [ ENABLE:%x ABOVE:%x/%x BELOW:%x/%x THRESH:%x ENERGY:%x ATTACK:%x DECAY:%x\n", + self->drce_state.enable, + self->drce_state.above.expand,self->drce_state.above.val, + self->drce_state.below.expand,self->drce_state.below.val, + self->drce_state.threshold, + self->drce_state.energy, + self->drce_state.attack, + self->drce_state.decay); + + printk("DRCE: reg [ %02x %02x %02x %02x %02x %02x ]\n", + (unsigned char)shadow[TAS3004_REG_DRC][0], + (unsigned char)shadow[TAS3004_REG_DRC][1], + (unsigned char)shadow[TAS3004_REG_DRC][2], + (unsigned char)shadow[TAS3004_REG_DRC][3], + (unsigned char)shadow[TAS3004_REG_DRC][4], + (unsigned char)shadow[TAS3004_REG_DRC][5]); +#endif + + return tas3004_sync_register(self, TAS3004_REG_DRC); +} + +static int +tas3004_drce_rw( struct tas3004_data_t *self, + u_int cmd, + u_long arg) +{ + int rc; + struct tas_drce_ctrl_t drce_ctrl; + void __user *argp = (void __user *)arg; + + if (copy_from_user(&drce_ctrl, argp, sizeof(struct tas_drce_ctrl_t))) + return -EFAULT; + +#ifdef DEBUG_DRCE + printk("DRCE: input [ FLAGS:%x ENABLE:%x ABOVE:%x/%x BELOW:%x/%x THRESH:%x ENERGY:%x ATTACK:%x DECAY:%x\n", + drce_ctrl.flags, + drce_ctrl.data.enable, + drce_ctrl.data.above.expand,drce_ctrl.data.above.val, + drce_ctrl.data.below.expand,drce_ctrl.data.below.val, + drce_ctrl.data.threshold, + drce_ctrl.data.energy, + drce_ctrl.data.attack, + drce_ctrl.data.decay); +#endif + + if (cmd & SIOC_IN) { + rc = tas3004_update_drce(self, drce_ctrl.flags, &drce_ctrl.data); + if (rc < 0) return rc; + } + + if (cmd & SIOC_OUT) { + if (drce_ctrl.flags & TAS_DRCE_ENABLE) + drce_ctrl.data.enable = self->drce_state.enable; + if (drce_ctrl.flags & TAS_DRCE_ABOVE_RATIO) + drce_ctrl.data.above = self->drce_state.above; + if (drce_ctrl.flags & TAS_DRCE_BELOW_RATIO) + drce_ctrl.data.below = self->drce_state.below; + if (drce_ctrl.flags & TAS_DRCE_THRESHOLD) + drce_ctrl.data.threshold = self->drce_state.threshold; + if (drce_ctrl.flags & TAS_DRCE_ENERGY) + drce_ctrl.data.energy = self->drce_state.energy; + if (drce_ctrl.flags & TAS_DRCE_ATTACK) + drce_ctrl.data.attack = self->drce_state.attack; + if (drce_ctrl.flags & TAS_DRCE_DECAY) + drce_ctrl.data.decay = self->drce_state.decay; + + if (copy_to_user(argp, &drce_ctrl, + sizeof(struct tas_drce_ctrl_t))) { + return -EFAULT; + } + } + + return 0; +} + +static void +tas3004_update_device_parameters(struct tas3004_data_t *self) +{ + char data; + int i; + + if (!self) return; + + if (self->output_id == TAS_OUTPUT_HEADPHONES) { + /* turn on allPass when headphones are plugged in */ + data = 0x02; + } else { + data = 0x00; + } + + tas3004_write_register(self, TAS3004_REG_MCR2, &data, WRITE_NORMAL | FORCE_WRITE); + + for (i=0; tas3004_eq_prefs[i]; i++) { + struct tas_eq_pref_t *eq = tas3004_eq_prefs[i]; + + if (eq->device_id == self->device_id && + (eq->output_id == 0 || eq->output_id == self->output_id) && + (eq->speaker_id == 0 || eq->speaker_id == self->speaker_id)) { + + tas3004_update_drce(self, TAS_DRCE_ALL, eq->drce); + tas3004_write_biquad_list(self, eq->filter_count, TAS_BIQUAD_FAST_LOAD, eq->biquads); + + break; + } + } +} + +static void +tas3004_device_change_handler(void *self) +{ + if (!self) return; + + tas3004_update_device_parameters((struct tas3004_data_t *)self); +} + +static struct work_struct device_change; + +static int +tas3004_output_device_change( struct tas3004_data_t *self, + int device_id, + int output_id, + int speaker_id) +{ + self->device_id=device_id; + self->output_id=output_id; + self->speaker_id=speaker_id; + + schedule_work(&device_change); + + return 0; +} + +static int +tas3004_device_ioctl( struct tas3004_data_t *self, + u_int cmd, + u_long arg) +{ + uint __user *argp = (void __user *)arg; + switch (cmd) { + case TAS_READ_EQ: + case TAS_WRITE_EQ: + return tas3004_eq_rw(self, cmd, arg); + + case TAS_READ_EQ_LIST: + case TAS_WRITE_EQ_LIST: + return tas3004_eq_list_rw(self, cmd, arg); + + case TAS_READ_EQ_FILTER_COUNT: + put_user(TAS3004_BIQUAD_FILTER_COUNT, argp); + return 0; + + case TAS_READ_EQ_CHANNEL_COUNT: + put_user(TAS3004_BIQUAD_CHANNEL_COUNT, argp); + return 0; + + case TAS_READ_DRCE: + case TAS_WRITE_DRCE: + return tas3004_drce_rw(self, cmd, arg); + + case TAS_READ_DRCE_CAPS: + put_user(TAS_DRCE_ENABLE | + TAS_DRCE_ABOVE_RATIO | + TAS_DRCE_BELOW_RATIO | + TAS_DRCE_THRESHOLD | + TAS_DRCE_ENERGY | + TAS_DRCE_ATTACK | + TAS_DRCE_DECAY, + argp); + return 0; + + case TAS_READ_DRCE_MIN: + case TAS_READ_DRCE_MAX: { + struct tas_drce_ctrl_t drce_ctrl; + const struct tas_drce_t *drce_copy; + + if (copy_from_user(&drce_ctrl, argp, + sizeof(struct tas_drce_ctrl_t))) { + return -EFAULT; + } + + if (cmd == TAS_READ_DRCE_MIN) { + drce_copy=&tas3004_drce_min; + } else { + drce_copy=&tas3004_drce_max; + } + + if (drce_ctrl.flags & TAS_DRCE_ABOVE_RATIO) { + drce_ctrl.data.above=drce_copy->above; + } + if (drce_ctrl.flags & TAS_DRCE_BELOW_RATIO) { + drce_ctrl.data.below=drce_copy->below; + } + if (drce_ctrl.flags & TAS_DRCE_THRESHOLD) { + drce_ctrl.data.threshold=drce_copy->threshold; + } + if (drce_ctrl.flags & TAS_DRCE_ENERGY) { + drce_ctrl.data.energy=drce_copy->energy; + } + if (drce_ctrl.flags & TAS_DRCE_ATTACK) { + drce_ctrl.data.attack=drce_copy->attack; + } + if (drce_ctrl.flags & TAS_DRCE_DECAY) { + drce_ctrl.data.decay=drce_copy->decay; + } + + if (copy_to_user(argp, &drce_ctrl, + sizeof(struct tas_drce_ctrl_t))) { + return -EFAULT; + } + } + } + + return -EINVAL; +} + +static int +tas3004_init_mixer(struct tas3004_data_t *self) +{ + unsigned char mcr = (1<<6)+(2<<4)+(2<<2); + + /* Make sure something answers on the i2c bus */ + if (tas3004_write_register(self, TAS3004_REG_MCR, &mcr, + WRITE_NORMAL | FORCE_WRITE) < 0) + return -1; + + tas3004_fast_load(self, 1); + + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD0); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD1); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD2); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD3); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD4); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD5); + (void)tas3004_sync_register(self,TAS3004_REG_RIGHT_BIQUAD6); + + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD0); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD1); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD2); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD3); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD4); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD5); + (void)tas3004_sync_register(self,TAS3004_REG_LEFT_BIQUAD6); + + tas3004_sync_register(self, TAS3004_REG_DRC); + + tas3004_sync_register(self, TAS3004_REG_MCR2); + + tas3004_fast_load(self, 0); + + tas3004_set_mixer_level(self, SOUND_MIXER_VOLUME, VOL_DEFAULT<<8 | VOL_DEFAULT); + tas3004_set_mixer_level(self, SOUND_MIXER_PCM, INPUT_DEFAULT<<8 | INPUT_DEFAULT); + tas3004_set_mixer_level(self, SOUND_MIXER_ALTPCM, 0); + tas3004_set_mixer_level(self, SOUND_MIXER_IMIX, 0); + + tas3004_set_mixer_level(self, SOUND_MIXER_BASS, BASS_DEFAULT); + tas3004_set_mixer_level(self, SOUND_MIXER_TREBLE, TREBLE_DEFAULT); + + tas3004_set_mixer_level(self, SOUND_MIXER_LINE,SW_INPUT_VOLUME_DEFAULT); + + return 0; +} + +static int +tas3004_uninit_mixer(struct tas3004_data_t *self) +{ + tas3004_set_mixer_level(self, SOUND_MIXER_VOLUME, 0); + tas3004_set_mixer_level(self, SOUND_MIXER_PCM, 0); + tas3004_set_mixer_level(self, SOUND_MIXER_ALTPCM, 0); + tas3004_set_mixer_level(self, SOUND_MIXER_IMIX, 0); + + tas3004_set_mixer_level(self, SOUND_MIXER_BASS, 0); + tas3004_set_mixer_level(self, SOUND_MIXER_TREBLE, 0); + + tas3004_set_mixer_level(self, SOUND_MIXER_LINE, 0); + + return 0; +} + +static int +tas3004_init(struct i2c_client *client) +{ + struct tas3004_data_t *self; + size_t sz = sizeof(*self) + (TAS3004_REG_MAX*sizeof(tas_shadow_t)); + char drce_init[] = { 0x69, 0x22, 0x9f, 0xb0, 0x60, 0xa0 }; + char mcr2 = 0; + int i, j; + + self = kmalloc(sz, GFP_KERNEL); + if (!self) + return -ENOMEM; + memset(self, 0, sz); + + self->super.client = client; + self->super.shadow = (tas_shadow_t *)(self+1); + self->output_id = TAS_OUTPUT_HEADPHONES; + + dev_set_drvdata(&client->dev, self); + + for (i = 0; i < TAS3004_BIQUAD_CHANNEL_COUNT; i++) + for (j = 0; j<TAS3004_BIQUAD_FILTER_COUNT; j++) + tas3004_write_biquad_shadow(self, i, j, + &tas3004_eq_unity); + + tas3004_write_register(self, TAS3004_REG_MCR2, &mcr2, WRITE_SHADOW); + tas3004_write_register(self, TAS3004_REG_DRC, drce_init, WRITE_SHADOW); + + INIT_WORK(&device_change, tas3004_device_change_handler, self); + return 0; +} + +static void +tas3004_uninit(struct tas3004_data_t *self) +{ + tas3004_uninit_mixer(self); + kfree(self); +} + + +struct tas_driver_hooks_t tas3004_hooks = { + .init = (tas_hook_init_t)tas3004_init, + .post_init = (tas_hook_post_init_t)tas3004_init_mixer, + .uninit = (tas_hook_uninit_t)tas3004_uninit, + .get_mixer_level = (tas_hook_get_mixer_level_t)tas3004_get_mixer_level, + .set_mixer_level = (tas_hook_set_mixer_level_t)tas3004_set_mixer_level, + .enter_sleep = (tas_hook_enter_sleep_t)tas3004_enter_sleep, + .leave_sleep = (tas_hook_leave_sleep_t)tas3004_leave_sleep, + .supported_mixers = (tas_hook_supported_mixers_t)tas3004_supported_mixers, + .mixer_is_stereo = (tas_hook_mixer_is_stereo_t)tas3004_mixer_is_stereo, + .stereo_mixers = (tas_hook_stereo_mixers_t)tas3004_stereo_mixers, + .output_device_change = (tas_hook_output_device_change_t)tas3004_output_device_change, + .device_ioctl = (tas_hook_device_ioctl_t)tas3004_device_ioctl +}; diff --git a/sound/oss/dmasound/tas3004.h b/sound/oss/dmasound/tas3004.h new file mode 100644 index 0000000..c6d584b --- /dev/null +++ b/sound/oss/dmasound/tas3004.h @@ -0,0 +1,77 @@ +/* + * Header file for the i2c/i2s based TA3004 sound chip used + * on some Apple hardware. Also known as "tumbler". + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive + * for more details. + * + * Written by Christopher C. Chimelis <chris@debian.org> + */ + +#ifndef _TAS3004_H_ +#define _TAS3004_H_ + +#include <linux/types.h> + +#include "tas_common.h" +#include "tas_eq_prefs.h" + +/* + * Macros that correspond to the registers that we write to + * when setting the various values. + */ + +#define TAS3004_VERSION "0.3" +#define TAS3004_DATE "20011214" + +#define I2C_DRIVERNAME_TAS3004 "TAS3004 driver V " TAS3004_VERSION +#define I2C_DRIVERID_TAS3004 (I2C_DRIVERID_TAS_BASE+1) + +extern struct tas_driver_hooks_t tas3004_hooks; +extern struct tas_gain_t tas3004_gain; +extern struct tas_eq_pref_t *tas3004_eq_prefs[]; + +enum tas3004_reg_t { + TAS3004_REG_MCR = 0x01, + TAS3004_REG_DRC = 0x02, + + TAS3004_REG_VOLUME = 0x04, + TAS3004_REG_TREBLE = 0x05, + TAS3004_REG_BASS = 0x06, + TAS3004_REG_LEFT_MIXER = 0x07, + TAS3004_REG_RIGHT_MIXER = 0x08, + + TAS3004_REG_LEFT_BIQUAD0 = 0x0a, + TAS3004_REG_LEFT_BIQUAD1 = 0x0b, + TAS3004_REG_LEFT_BIQUAD2 = 0x0c, + TAS3004_REG_LEFT_BIQUAD3 = 0x0d, + TAS3004_REG_LEFT_BIQUAD4 = 0x0e, + TAS3004_REG_LEFT_BIQUAD5 = 0x0f, + TAS3004_REG_LEFT_BIQUAD6 = 0x10, + + TAS3004_REG_RIGHT_BIQUAD0 = 0x13, + TAS3004_REG_RIGHT_BIQUAD1 = 0x14, + TAS3004_REG_RIGHT_BIQUAD2 = 0x15, + TAS3004_REG_RIGHT_BIQUAD3 = 0x16, + TAS3004_REG_RIGHT_BIQUAD4 = 0x17, + TAS3004_REG_RIGHT_BIQUAD5 = 0x18, + TAS3004_REG_RIGHT_BIQUAD6 = 0x19, + + TAS3004_REG_LEFT_LOUD_BIQUAD = 0x21, + TAS3004_REG_RIGHT_LOUD_BIQUAD = 0x22, + + TAS3004_REG_LEFT_LOUD_BIQUAD_GAIN = 0x23, + TAS3004_REG_RIGHT_LOUD_BIQUAD_GAIN = 0x24, + + TAS3004_REG_TEST = 0x29, + + TAS3004_REG_ANALOG_CTRL = 0x40, + TAS3004_REG_TEST1 = 0x41, + TAS3004_REG_TEST2 = 0x42, + TAS3004_REG_MCR2 = 0x43, + + TAS3004_REG_MAX = 0x44 +}; + +#endif /* _TAS3004_H_ */ diff --git a/sound/oss/dmasound/tas3004_tables.c b/sound/oss/dmasound/tas3004_tables.c new file mode 100644 index 0000000..b910e0a --- /dev/null +++ b/sound/oss/dmasound/tas3004_tables.c @@ -0,0 +1,301 @@ +#include "tas3004.h" +#include "tas_eq_prefs.h" + +static struct tas_drce_t eqp_17_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -19.12 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_17_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0fd0d4, 0xe05e56, 0x0fd0d4, 0xe05ee1, 0x0fa234 } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x0910d7, 0x088e1a, 0x030651, 0x01dcb1, 0x02c892 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0ff895, 0xe0970b, 0x0f7f00, 0xe0970b, 0x0f7795 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0fd1c4, 0xe1ac22, 0x0ec8cf, 0xe1ac22, 0x0e9a94 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x0f7c1c, 0xe3cc03, 0x0df786, 0xe3cc03, 0x0d73a2 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x11fb92, 0xf5a1a0, 0x073cd2, 0xf5a1a0, 0x093865 } } }, + { .channel = 0, .filter = 6, .data = { .coeff = { 0x0e17a9, 0x068b6c, 0x08a0e5, 0x068b6c, 0x06b88e } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0fd0d4, 0xe05e56, 0x0fd0d4, 0xe05ee1, 0x0fa234 } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x0910d7, 0x088e1a, 0x030651, 0x01dcb1, 0x02c892 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0ff895, 0xe0970b, 0x0f7f00, 0xe0970b, 0x0f7795 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0fd1c4, 0xe1ac22, 0x0ec8cf, 0xe1ac22, 0x0e9a94 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x0f7c1c, 0xe3cc03, 0x0df786, 0xe3cc03, 0x0d73a2 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x11fb92, 0xf5a1a0, 0x073cd2, 0xf5a1a0, 0x093865 } } }, + { .channel = 1, .filter = 6, .data = { .coeff = { 0x0e17a9, 0x068b6c, 0x08a0e5, 0x068b6c, 0x06b88e } } } +}; + +static struct tas_eq_pref_t eqp_17_1_0 = { + .sample_rate = 44100, + .device_id = 0x17, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_17_1_0_drce, + + .filter_count = 14, + .biquads = eqp_17_1_0_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_18_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -13.14 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_18_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0f5514, 0xe155d7, 0x0f5514, 0xe15cfa, 0x0eb14b } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x06ec33, 0x02abe3, 0x015eef, 0xf764d9, 0x03922d } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0ef5f2, 0xe67d1f, 0x0bcf37, 0xe67d1f, 0x0ac529 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0db050, 0xe5be4d, 0x0d0c78, 0xe5be4d, 0x0abcc8 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x0f1298, 0xe64ec6, 0x0cc03e, 0xe64ec6, 0x0bd2d7 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0c641a, 0x06537a, 0x08d155, 0x06537a, 0x053570 } } }, + { .channel = 0, .filter = 6, .data = { .coeff = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0f5514, 0xe155d7, 0x0f5514, 0xe15cfa, 0x0eb14b } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x06ec33, 0x02abe3, 0x015eef, 0xf764d9, 0x03922d } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0ef5f2, 0xe67d1f, 0x0bcf37, 0xe67d1f, 0x0ac529 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0db050, 0xe5be4d, 0x0d0c78, 0xe5be4d, 0x0abcc8 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x0f1298, 0xe64ec6, 0x0cc03e, 0xe64ec6, 0x0bd2d7 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0c641a, 0x06537a, 0x08d155, 0x06537a, 0x053570 } } }, + { .channel = 1, .filter = 6, .data = { .coeff = { 0x100000, 0x000000, 0x000000, 0x000000, 0x000000 } } } +}; + +static struct tas_eq_pref_t eqp_18_1_0 = { + .sample_rate = 44100, + .device_id = 0x18, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_18_1_0_drce, + + .filter_count = 14, + .biquads = eqp_18_1_0_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_1a_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -10.75 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_1a_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0fb8fd, 0xe08e04, 0x0fb8fd, 0xe08f40, 0x0f7336 } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x06371d, 0x0c6e3a, 0x06371d, 0x05bfd3, 0x031ca2 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0fa1c0, 0xe18692, 0x0f030e, 0xe18692, 0x0ea4ce } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0fe495, 0xe17eff, 0x0f0452, 0xe17eff, 0x0ee8e7 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x100857, 0xe7e71c, 0x0e9599, 0xe7e71c, 0x0e9df1 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0fb26e, 0x06a82c, 0x0db2b4, 0x06a82c, 0x0d6522 } } }, + { .channel = 0, .filter = 6, .data = { .coeff = { 0x11419d, 0xf06cbf, 0x0a4f6e, 0xf06cbf, 0x0b910c } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0fb8fd, 0xe08e04, 0x0fb8fd, 0xe08f40, 0x0f7336 } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x06371d, 0x0c6e3a, 0x06371d, 0x05bfd3, 0x031ca2 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0fa1c0, 0xe18692, 0x0f030e, 0xe18692, 0x0ea4ce } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0fe495, 0xe17eff, 0x0f0452, 0xe17eff, 0x0ee8e7 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x100857, 0xe7e71c, 0x0e9599, 0xe7e71c, 0x0e9df1 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0fb26e, 0x06a82c, 0x0db2b4, 0x06a82c, 0x0d6522 } } }, + { .channel = 1, .filter = 6, .data = { .coeff = { 0x11419d, 0xf06cbf, 0x0a4f6e, 0xf06cbf, 0x0b910c } } } +}; + +static struct tas_eq_pref_t eqp_1a_1_0 = { + .sample_rate = 44100, + .device_id = 0x1a, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_1a_1_0_drce, + + .filter_count = 14, + .biquads = eqp_1a_1_0_biquads +}; + +/* ======================================================================== */ + +static struct tas_drce_t eqp_1c_1_0_drce={ + .enable = 1, + .above = { .val = 3.0 * (1<<8), .expand = 0 }, + .below = { .val = 1.0 * (1<<8), .expand = 0 }, + .threshold = -14.34 * (1<<8), + .energy = 2.4 * (1<<12), + .attack = 0.013 * (1<<12), + .decay = 0.212 * (1<<12), +}; + +static struct tas_biquad_ctrl_t eqp_1c_1_0_biquads[]={ + { .channel = 0, .filter = 0, .data = { .coeff = { 0x0f4f95, 0xe160d4, 0x0f4f95, 0xe1686e, 0x0ea6c5 } } }, + { .channel = 0, .filter = 1, .data = { .coeff = { 0x066b92, 0x0290d4, 0x0148a0, 0xf6853f, 0x03bfc7 } } }, + { .channel = 0, .filter = 2, .data = { .coeff = { 0x0f57dc, 0xe51c91, 0x0dd1cb, 0xe51c91, 0x0d29a8 } } }, + { .channel = 0, .filter = 3, .data = { .coeff = { 0x0df1cb, 0xe4fa84, 0x0d7cdc, 0xe4fa84, 0x0b6ea7 } } }, + { .channel = 0, .filter = 4, .data = { .coeff = { 0x0eba36, 0xe6aa48, 0x0b9f52, 0xe6aa48, 0x0a5989 } } }, + { .channel = 0, .filter = 5, .data = { .coeff = { 0x0caf02, 0x05ef9d, 0x084beb, 0x05ef9d, 0x04faee } } }, + { .channel = 0, .filter = 6, .data = { .coeff = { 0x0fc686, 0xe22947, 0x0e4b5d, 0xe22947, 0x0e11e4 } } }, + + { .channel = 1, .filter = 0, .data = { .coeff = { 0x0f4f95, 0xe160d4, 0x0f4f95, 0xe1686e, 0x0ea6c5 } } }, + { .channel = 1, .filter = 1, .data = { .coeff = { 0x066b92, 0x0290d4, 0x0148a0, 0xf6853f, 0x03bfc7 } } }, + { .channel = 1, .filter = 2, .data = { .coeff = { 0x0f57dc, 0xe51c91, 0x0dd1cb, 0xe51c91, 0x0d29a8 } } }, + { .channel = 1, .filter = 3, .data = { .coeff = { 0x0df1cb, 0xe4fa84, 0x0d7cdc, 0xe4fa84, 0x0b6ea7 } } }, + { .channel = 1, .filter = 4, .data = { .coeff = { 0x0eba36, 0xe6aa48, 0x0b9f52, 0xe6aa48, 0x0a5989 } } }, + { .channel = 1, .filter = 5, .data = { .coeff = { 0x0caf02, 0x05ef9d, 0x084beb, 0x05ef9d, 0x04faee } } }, + { .channel = 1, .filter = 6, .data = { .coeff = { 0x0fc686, 0xe22947, 0x0e4b5d, 0xe22947, 0x0e11e4 } } } +}; + +static struct tas_eq_pref_t eqp_1c_1_0 = { + .sample_rate = 44100, + .device_id = 0x1c, + .output_id = TAS_OUTPUT_INTERNAL_SPKR, + .speaker_id = 0x00, + + .drce = &eqp_1c_1_0_drce, + + .filter_count = 14, + .biquads = eqp_1c_1_0_biquads +}; + +/* ======================================================================== */ + +static uint tas3004_master_tab[]={ + 0x0, 0x75, 0x9c, 0xbb, + 0xdb, 0xfb, 0x11e, 0x143, + 0x16b, 0x196, 0x1c3, 0x1f5, + 0x229, 0x263, 0x29f, 0x2e1, + 0x328, 0x373, 0x3c5, 0x41b, + 0x478, 0x4dc, 0x547, 0x5b8, + 0x633, 0x6b5, 0x740, 0x7d5, + 0x873, 0x91c, 0x9d2, 0xa92, + 0xb5e, 0xc39, 0xd22, 0xe19, + 0xf20, 0x1037, 0x1161, 0x129e, + 0x13ed, 0x1551, 0x16ca, 0x185d, + 0x1a08, 0x1bcc, 0x1dac, 0x1fa7, + 0x21c1, 0x23fa, 0x2655, 0x28d6, + 0x2b7c, 0x2e4a, 0x3141, 0x3464, + 0x37b4, 0x3b35, 0x3ee9, 0x42d3, + 0x46f6, 0x4b53, 0x4ff0, 0x54ce, + 0x59f2, 0x5f5f, 0x6519, 0x6b24, + 0x7183, 0x783c, 0x7f53, 0x86cc, + 0x8ead, 0x96fa, 0x9fba, 0xa8f2, + 0xb2a7, 0xbce1, 0xc7a5, 0xd2fa, + 0xdee8, 0xeb75, 0xf8aa, 0x1068e, + 0x1152a, 0x12487, 0x134ad, 0x145a5, + 0x1577b, 0x16a37, 0x17df5, 0x192bd, + 0x1a890, 0x1bf7b, 0x1d78d, 0x1f0d1, + 0x20b55, 0x22727, 0x24456, 0x262f2, + 0x2830b +}; + +static uint tas3004_mixer_tab[]={ + 0x0, 0x748, 0x9be, 0xbaf, + 0xda4, 0xfb1, 0x11de, 0x1431, + 0x16ad, 0x1959, 0x1c37, 0x1f4b, + 0x2298, 0x2628, 0x29fb, 0x2e12, + 0x327d, 0x3734, 0x3c47, 0x41b4, + 0x4787, 0x4dbe, 0x546d, 0x5b86, + 0x632e, 0x6b52, 0x7400, 0x7d54, + 0x873b, 0x91c6, 0x9d1a, 0xa920, + 0xb5e5, 0xc38c, 0xd21b, 0xe18f, + 0xf1f5, 0x1036a, 0x1160f, 0x129d6, + 0x13ed0, 0x1550c, 0x16ca0, 0x185c9, + 0x1a07b, 0x1bcc3, 0x1dab9, 0x1fa75, + 0x21c0f, 0x23fa3, 0x26552, 0x28d64, + 0x2b7c9, 0x2e4a2, 0x31411, 0x3463b, + 0x37b44, 0x3b353, 0x3ee94, 0x42d30, + 0x46f55, 0x4b533, 0x4fefc, 0x54ce5, + 0x59f25, 0x5f5f6, 0x65193, 0x6b23c, + 0x71835, 0x783c3, 0x7f52c, 0x86cc0, + 0x8eacc, 0x96fa5, 0x9fba0, 0xa8f1a, + 0xb2a71, 0xbce0a, 0xc7a4a, 0xd2fa0, + 0xdee7b, 0xeb752, 0xf8a9f, 0x1068e4, + 0x1152a3, 0x12486a, 0x134ac8, 0x145a55, + 0x1577ac, 0x16a370, 0x17df51, 0x192bc2, + 0x1a88f8, 0x1bf7b7, 0x1d78c9, 0x1f0d04, + 0x20b542, 0x227268, 0x244564, 0x262f26, + 0x2830af +}; + +static uint tas3004_treble_tab[]={ + 0x96, 0x95, 0x95, 0x94, + 0x93, 0x92, 0x92, 0x91, + 0x90, 0x90, 0x8f, 0x8e, + 0x8d, 0x8d, 0x8c, 0x8b, + 0x8a, 0x8a, 0x89, 0x88, + 0x88, 0x87, 0x86, 0x85, + 0x85, 0x84, 0x83, 0x83, + 0x82, 0x81, 0x80, 0x80, + 0x7f, 0x7e, 0x7e, 0x7d, + 0x7c, 0x7b, 0x7b, 0x7a, + 0x79, 0x78, 0x78, 0x77, + 0x76, 0x76, 0x75, 0x74, + 0x73, 0x73, 0x72, 0x71, + 0x71, 0x68, 0x45, 0x5b, + 0x6d, 0x6c, 0x6b, 0x6a, + 0x69, 0x68, 0x67, 0x66, + 0x65, 0x63, 0x62, 0x62, + 0x60, 0x5e, 0x5c, 0x5b, + 0x59, 0x57, 0x55, 0x53, + 0x52, 0x4f, 0x4d, 0x4a, + 0x48, 0x46, 0x43, 0x40, + 0x3d, 0x3a, 0x36, 0x33, + 0x2f, 0x2c, 0x27, 0x23, + 0x1f, 0x1a, 0x15, 0xf, + 0x8, 0x5, 0x2, 0x1, + 0x1 +}; + +static uint tas3004_bass_tab[]={ + 0x96, 0x95, 0x95, 0x94, + 0x93, 0x92, 0x92, 0x91, + 0x90, 0x90, 0x8f, 0x8e, + 0x8d, 0x8d, 0x8c, 0x8b, + 0x8a, 0x8a, 0x89, 0x88, + 0x88, 0x87, 0x86, 0x85, + 0x85, 0x84, 0x83, 0x83, + 0x82, 0x81, 0x80, 0x80, + 0x7f, 0x7e, 0x7e, 0x7d, + 0x7c, 0x7b, 0x7b, 0x7a, + 0x79, 0x78, 0x78, 0x77, + 0x76, 0x76, 0x75, 0x74, + 0x73, 0x73, 0x72, 0x71, + 0x70, 0x6f, 0x6e, 0x6d, + 0x6c, 0x6b, 0x6a, 0x6a, + 0x69, 0x67, 0x66, 0x66, + 0x65, 0x63, 0x62, 0x62, + 0x61, 0x60, 0x5e, 0x5d, + 0x5b, 0x59, 0x57, 0x55, + 0x53, 0x51, 0x4f, 0x4c, + 0x4a, 0x48, 0x46, 0x44, + 0x41, 0x3e, 0x3b, 0x38, + 0x36, 0x33, 0x2f, 0x2b, + 0x28, 0x24, 0x20, 0x1c, + 0x17, 0x12, 0xd, 0x7, + 0x1 +}; + +struct tas_gain_t tas3004_gain={ + .master = tas3004_master_tab, + .treble = tas3004_treble_tab, + .bass = tas3004_bass_tab, + .mixer = tas3004_mixer_tab +}; + +struct tas_eq_pref_t *tas3004_eq_prefs[]={ + &eqp_17_1_0, + &eqp_18_1_0, + &eqp_1a_1_0, + &eqp_1c_1_0, + NULL +}; diff --git a/sound/oss/dmasound/tas_common.c b/sound/oss/dmasound/tas_common.c new file mode 100644 index 0000000..d36a1fe --- /dev/null +++ b/sound/oss/dmasound/tas_common.c @@ -0,0 +1,214 @@ +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/proc_fs.h> +#include <linux/ioport.h> +#include <linux/sysctl.h> +#include <linux/types.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <asm/uaccess.h> +#include <asm/errno.h> +#include <asm/io.h> +#include <asm/prom.h> + +#include "tas_common.h" + +#define CALL0(proc) \ + do { \ + struct tas_data_t *self; \ + if (!tas_client || driver_hooks == NULL) \ + return -1; \ + self = dev_get_drvdata(&tas_client->dev); \ + if (driver_hooks->proc) \ + return driver_hooks->proc(self); \ + else \ + return -EINVAL; \ + } while (0) + +#define CALL(proc,arg...) \ + do { \ + struct tas_data_t *self; \ + if (!tas_client || driver_hooks == NULL) \ + return -1; \ + self = dev_get_drvdata(&tas_client->dev); \ + if (driver_hooks->proc) \ + return driver_hooks->proc(self, ## arg); \ + else \ + return -EINVAL; \ + } while (0) + + +static u8 tas_i2c_address = 0x34; +static struct i2c_client *tas_client; +static struct device_node* tas_node; + +static int tas_attach_adapter(struct i2c_adapter *); +static int tas_detach_client(struct i2c_client *); + +struct i2c_driver tas_driver = { + .owner = THIS_MODULE, + .name = "tas", + .flags = I2C_DF_NOTIFY, + .attach_adapter = tas_attach_adapter, + .detach_client = tas_detach_client, +}; + +struct tas_driver_hooks_t *driver_hooks; + +int +tas_register_driver(struct tas_driver_hooks_t *hooks) +{ + driver_hooks = hooks; + return 0; +} + +int +tas_get_mixer_level(int mixer, uint *level) +{ + CALL(get_mixer_level,mixer,level); +} + +int +tas_set_mixer_level(int mixer,uint level) +{ + CALL(set_mixer_level,mixer,level); +} + +int +tas_enter_sleep(void) +{ + CALL0(enter_sleep); +} + +int +tas_leave_sleep(void) +{ + CALL0(leave_sleep); +} + +int +tas_supported_mixers(void) +{ + CALL0(supported_mixers); +} + +int +tas_mixer_is_stereo(int mixer) +{ + CALL(mixer_is_stereo,mixer); +} + +int +tas_stereo_mixers(void) +{ + CALL0(stereo_mixers); +} + +int +tas_output_device_change(int device_id,int layout_id,int speaker_id) +{ + CALL(output_device_change,device_id,layout_id,speaker_id); +} + +int +tas_device_ioctl(u_int cmd, u_long arg) +{ + CALL(device_ioctl,cmd,arg); +} + +int +tas_post_init(void) +{ + CALL0(post_init); +} + +static int +tas_detect_client(struct i2c_adapter *adapter, int address) +{ + static const char *client_name = "tas Digital Equalizer"; + struct i2c_client *new_client; + int rc = -ENODEV; + + if (!driver_hooks) { + printk(KERN_ERR "tas_detect_client called with no hooks !\n"); + return -ENODEV; + } + + new_client = kmalloc(sizeof(*new_client), GFP_KERNEL); + if (!new_client) + return -ENOMEM; + memset(new_client, 0, sizeof(*new_client)); + + new_client->addr = address; + new_client->adapter = adapter; + new_client->driver = &tas_driver; + strlcpy(new_client->name, client_name, DEVICE_NAME_SIZE); + + if (driver_hooks->init(new_client)) + goto bail; + + /* Tell the i2c layer a new client has arrived */ + if (i2c_attach_client(new_client)) { + driver_hooks->uninit(dev_get_drvdata(&new_client->dev)); + goto bail; + } + + tas_client = new_client; + return 0; + bail: + tas_client = NULL; + kfree(new_client); + return rc; +} + +static int +tas_attach_adapter(struct i2c_adapter *adapter) +{ + if (!strncmp(adapter->name, "mac-io", 6)) + return tas_detect_client(adapter, tas_i2c_address); + return 0; +} + +static int +tas_detach_client(struct i2c_client *client) +{ + if (client == tas_client) { + driver_hooks->uninit(dev_get_drvdata(&client->dev)); + + i2c_detach_client(client); + kfree(client); + } + return 0; +} + +void +tas_cleanup(void) +{ + i2c_del_driver(&tas_driver); +} + +int __init +tas_init(int driver_id, const char *driver_name) +{ + u32* paddr; + + printk(KERN_INFO "tas driver [%s])\n", driver_name); + +#ifndef CONFIG_I2C_KEYWEST + request_module("i2c-keywest"); +#endif + tas_node = find_devices("deq"); + if (tas_node == NULL) + return -ENODEV; + paddr = (u32 *)get_property(tas_node, "i2c-address", NULL); + if (paddr) { + tas_i2c_address = (*paddr) >> 1; + printk(KERN_INFO "using i2c address: 0x%x from device-tree\n", + tas_i2c_address); + } else + printk(KERN_INFO "using i2c address: 0x%x (default)\n", + tas_i2c_address); + + return i2c_add_driver(&tas_driver); +} diff --git a/sound/oss/dmasound/tas_common.h b/sound/oss/dmasound/tas_common.h new file mode 100644 index 0000000..3a6d486 --- /dev/null +++ b/sound/oss/dmasound/tas_common.h @@ -0,0 +1,284 @@ +#ifndef _TAS_COMMON_H_ +#define _TAS_COMMON_H_ + +#include <linux/i2c.h> +#include <linux/soundcard.h> +#include <asm/string.h> + +#define I2C_DRIVERID_TAS_BASE (0xFEBA) + +#define SET_4_20(shadow, offset, val) \ + do { \ + (shadow)[(offset)+0] = ((val) >> 16) & 0xff; \ + (shadow)[(offset)+1] = ((val) >> 8) & 0xff; \ + (shadow)[(offset)+2] = ((val) >> 0) & 0xff; \ + } while (0) + +#define GET_4_20(shadow, offset) \ + (((u_int)((shadow)[(offset)+0]) << 16) | \ + ((u_int)((shadow)[(offset)+1]) << 8) | \ + ((u_int)((shadow)[(offset)+2]) << 0)) + + +#define TAS_BIQUAD_FAST_LOAD 0x01 + +#define TAS_DRCE_ENABLE 0x01 +#define TAS_DRCE_ABOVE_RATIO 0x02 +#define TAS_DRCE_BELOW_RATIO 0x04 +#define TAS_DRCE_THRESHOLD 0x08 +#define TAS_DRCE_ENERGY 0x10 +#define TAS_DRCE_ATTACK 0x20 +#define TAS_DRCE_DECAY 0x40 + +#define TAS_DRCE_ALL 0x7f + + +#define TAS_OUTPUT_HEADPHONES 0x00 +#define TAS_OUTPUT_INTERNAL_SPKR 0x01 +#define TAS_OUTPUT_EXTERNAL_SPKR 0x02 + + +union tas_biquad_t { + struct { + int b0,b1,b2,a1,a2; + } coeff; + int buf[5]; +}; + +struct tas_biquad_ctrl_t { + u_int channel:4; + u_int filter:4; + + union tas_biquad_t data; +}; + +struct tas_biquad_ctrl_list_t { + int flags; + int filter_count; + struct tas_biquad_ctrl_t biquads[0]; +}; + +struct tas_ratio_t { + unsigned short val; /* 8.8 */ + unsigned short expand; /* 0 = compress, !0 = expand. */ +}; + +struct tas_drce_t { + unsigned short enable; + struct tas_ratio_t above; + struct tas_ratio_t below; + short threshold; /* dB, 8.8 signed */ + unsigned short energy; /* seconds, 4.12 unsigned */ + unsigned short attack; /* seconds, 4.12 unsigned */ + unsigned short decay; /* seconds, 4.12 unsigned */ +}; + +struct tas_drce_ctrl_t { + uint flags; + + struct tas_drce_t data; +}; + +struct tas_gain_t +{ + unsigned int *master; + unsigned int *treble; + unsigned int *bass; + unsigned int *mixer; +}; + +typedef char tas_shadow_t[0x45]; + +struct tas_data_t +{ + struct i2c_client *client; + tas_shadow_t *shadow; + uint mixer[SOUND_MIXER_NRDEVICES]; +}; + +typedef int (*tas_hook_init_t)(struct i2c_client *); +typedef int (*tas_hook_post_init_t)(struct tas_data_t *); +typedef void (*tas_hook_uninit_t)(struct tas_data_t *); + +typedef int (*tas_hook_get_mixer_level_t)(struct tas_data_t *,int,uint *); +typedef int (*tas_hook_set_mixer_level_t)(struct tas_data_t *,int,uint); + +typedef int (*tas_hook_enter_sleep_t)(struct tas_data_t *); +typedef int (*tas_hook_leave_sleep_t)(struct tas_data_t *); + +typedef int (*tas_hook_supported_mixers_t)(struct tas_data_t *); +typedef int (*tas_hook_mixer_is_stereo_t)(struct tas_data_t *,int); +typedef int (*tas_hook_stereo_mixers_t)(struct tas_data_t *); + +typedef int (*tas_hook_output_device_change_t)(struct tas_data_t *,int,int,int); +typedef int (*tas_hook_device_ioctl_t)(struct tas_data_t *,u_int,u_long); + +struct tas_driver_hooks_t { + /* + * All hardware initialisation must be performed in + * post_init(), as tas_dmasound_init() does a hardware reset. + * + * init() is called before tas_dmasound_init() so that + * ouput_device_change() is always called after i2c driver + * initialisation. The implication is that + * output_device_change() must cope with the fact that it + * may be called before post_init(). + */ + + tas_hook_init_t init; + tas_hook_post_init_t post_init; + tas_hook_uninit_t uninit; + + tas_hook_get_mixer_level_t get_mixer_level; + tas_hook_set_mixer_level_t set_mixer_level; + + tas_hook_enter_sleep_t enter_sleep; + tas_hook_leave_sleep_t leave_sleep; + + tas_hook_supported_mixers_t supported_mixers; + tas_hook_mixer_is_stereo_t mixer_is_stereo; + tas_hook_stereo_mixers_t stereo_mixers; + + tas_hook_output_device_change_t output_device_change; + tas_hook_device_ioctl_t device_ioctl; +}; + +enum tas_write_mode_t { + WRITE_HW = 0x01, + WRITE_SHADOW = 0x02, + WRITE_NORMAL = 0x03, + FORCE_WRITE = 0x04 +}; + +static inline uint +tas_mono_to_stereo(uint mono) +{ + mono &=0xff; + return mono | (mono<<8); +} + +/* + * Todo: make these functions a bit more efficient ! + */ +static inline int +tas_write_register( struct tas_data_t *self, + uint reg_num, + uint reg_width, + char *data, + uint write_mode) +{ + int rc; + + if (reg_width==0 || data==NULL || self==NULL) + return -EINVAL; + if (!(write_mode & FORCE_WRITE) && + !memcmp(data,self->shadow[reg_num],reg_width)) + return 0; + + if (write_mode & WRITE_SHADOW) + memcpy(self->shadow[reg_num],data,reg_width); + if (write_mode & WRITE_HW) { + rc=i2c_smbus_write_block_data(self->client, + reg_num, + reg_width, + data); + if (rc < 0) { + printk("tas: I2C block write failed \n"); + return rc; + } + } + return 0; +} + +static inline int +tas_sync_register( struct tas_data_t *self, + uint reg_num, + uint reg_width) +{ + int rc; + + if (reg_width==0 || self==NULL) + return -EINVAL; + rc=i2c_smbus_write_block_data(self->client, + reg_num, + reg_width, + self->shadow[reg_num]); + if (rc < 0) { + printk("tas: I2C block write failed \n"); + return rc; + } + return 0; +} + +static inline int +tas_write_byte_register( struct tas_data_t *self, + uint reg_num, + char data, + uint write_mode) +{ + if (self==NULL) + return -1; + if (!(write_mode & FORCE_WRITE) && data != self->shadow[reg_num][0]) + return 0; + if (write_mode & WRITE_SHADOW) + self->shadow[reg_num][0]=data; + if (write_mode & WRITE_HW) { + if (i2c_smbus_write_byte_data(self->client, reg_num, data) < 0) { + printk("tas: I2C byte write failed \n"); + return -1; + } + } + return 0; +} + +static inline int +tas_sync_byte_register( struct tas_data_t *self, + uint reg_num, + uint reg_width) +{ + if (reg_width==0 || self==NULL) + return -1; + if (i2c_smbus_write_byte_data( + self->client, reg_num, self->shadow[reg_num][0]) < 0) { + printk("tas: I2C byte write failed \n"); + return -1; + } + return 0; +} + +static inline int +tas_read_register( struct tas_data_t *self, + uint reg_num, + uint reg_width, + char *data) +{ + if (reg_width==0 || data==NULL || self==NULL) + return -1; + memcpy(data,self->shadow[reg_num],reg_width); + return 0; +} + +extern int tas_register_driver(struct tas_driver_hooks_t *hooks); + +extern int tas_get_mixer_level(int mixer,uint *level); +extern int tas_set_mixer_level(int mixer,uint level); +extern int tas_enter_sleep(void); +extern int tas_leave_sleep(void); +extern int tas_supported_mixers(void); +extern int tas_mixer_is_stereo(int mixer); +extern int tas_stereo_mixers(void); +extern int tas_output_device_change(int,int,int); +extern int tas_device_ioctl(u_int, u_long); + +extern void tas_cleanup(void); +extern int tas_init(int driver_id,const char *driver_name); +extern int tas_post_init(void); + +#endif /* _TAS_COMMON_H_ */ +/* + * Local Variables: + * tab-width: 8 + * indent-tabs-mode: t + * c-basic-offset: 8 + * End: + */ diff --git a/sound/oss/dmasound/tas_eq_prefs.h b/sound/oss/dmasound/tas_eq_prefs.h new file mode 100644 index 0000000..3a994ed --- /dev/null +++ b/sound/oss/dmasound/tas_eq_prefs.h @@ -0,0 +1,24 @@ +#ifndef _TAS_EQ_PREFS_H_ +#define _TAS_EQ_PREFS_H_ + +struct tas_eq_pref_t { + u_int sample_rate; + u_int device_id; + u_int output_id; + u_int speaker_id; + + struct tas_drce_t *drce; + + u_int filter_count; + struct tas_biquad_ctrl_t *biquads; +}; + +#endif /* _TAS_EQ_PREFS_H_ */ + +/* + * Local Variables: + * tab-width: 8 + * indent-tabs-mode: t + * c-basic-offset: 8 + * End: + */ diff --git a/sound/oss/dmasound/tas_ioctl.h b/sound/oss/dmasound/tas_ioctl.h new file mode 100644 index 0000000..dccae3a --- /dev/null +++ b/sound/oss/dmasound/tas_ioctl.h @@ -0,0 +1,24 @@ +#ifndef _TAS_IOCTL_H_ +#define _TAS_IOCTL_H_ + +#include <linux/i2c.h> +#include <linux/soundcard.h> + + +#define TAS_READ_EQ _SIOR('t',0,struct tas_biquad_ctrl_t) +#define TAS_WRITE_EQ _SIOW('t',0,struct tas_biquad_ctrl_t) + +#define TAS_READ_EQ_LIST _SIOR('t',1,struct tas_biquad_ctrl_t) +#define TAS_WRITE_EQ_LIST _SIOW('t',1,struct tas_biquad_ctrl_t) + +#define TAS_READ_EQ_FILTER_COUNT _SIOR('t',2,int) +#define TAS_READ_EQ_CHANNEL_COUNT _SIOR('t',3,int) + +#define TAS_READ_DRCE _SIOR('t',4,struct tas_drce_ctrl_t) +#define TAS_WRITE_DRCE _SIOW('t',4,struct tas_drce_ctrl_t) + +#define TAS_READ_DRCE_CAPS _SIOR('t',5,int) +#define TAS_READ_DRCE_MIN _SIOR('t',6,int) +#define TAS_READ_DRCE_MAX _SIOR('t',7,int) + +#endif diff --git a/sound/oss/dmasound/trans_16.c b/sound/oss/dmasound/trans_16.c new file mode 100644 index 0000000..23562e9 --- /dev/null +++ b/sound/oss/dmasound/trans_16.c @@ -0,0 +1,897 @@ +/* + * linux/sound/oss/dmasound/trans_16.c + * + * 16 bit translation routines. Only used by Power mac at present. + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and + * history prior to 08/02/2001. + * + * 08/02/2001 Iain Sandoe + * split from dmasound_awacs.c + * 11/29/2003 Renzo Davoli (King Enzo) + * - input resampling (for soft rate < hard rate) + * - software line in gain control + */ + +#include <linux/soundcard.h> +#include <asm/uaccess.h> +#include "dmasound.h" + +static short dmasound_alaw2dma16[] ; +static short dmasound_ulaw2dma16[] ; + +static ssize_t pmac_ct_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ct_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ct_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ct_s16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ct_u16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); + +static ssize_t pmac_ctx_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ctx_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ctx_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ctx_s16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ctx_u16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); + +static ssize_t pmac_ct_s16_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t pmac_ct_u16_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); + +/*** Translations ************************************************************/ + +static int expand_data; /* Data for expanding */ + +static ssize_t pmac_ct_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + short *table = dmasound.soft.format == AFMT_MU_LAW + ? dmasound_ulaw2dma16 : dmasound_alaw2dma16; + ssize_t count, used; + short *p = (short *) &frame[*frameUsed]; + int val, stereo = dmasound.soft.stereo; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + used = count = min_t(unsigned long, 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 pmac_ct_s8(const u_char __user *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 = dmasound.soft.stereo; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + used = count = min_t(unsigned long, 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 pmac_ct_u8(const u_char __user *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 = dmasound.soft.stereo; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + used = count = min_t(unsigned long, 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; +} + + +static ssize_t pmac_ct_s16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + int stereo = dmasound.soft.stereo; + short *fp = (short *) &frame[*frameUsed]; + + frameLeft >>= 2; + userCount >>= (stereo? 2: 1); + used = count = min_t(unsigned long, userCount, frameLeft); + if (!stereo) { + short __user *up = (short __user *) 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 pmac_ct_u16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + int mask = (dmasound.soft.format == AFMT_U16_LE? 0x0080: 0x8000); + int stereo = dmasound.soft.stereo; + short *fp = (short *) &frame[*frameUsed]; + short __user *up = (short __user *) userPtr; + + frameLeft >>= 2; + userCount >>= (stereo? 2: 1); + used = count = min_t(unsigned long, userCount, frameLeft); + while (count > 0) { + short 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 pmac_ctx_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + unsigned short *table = (unsigned short *) + (dmasound.soft.format == AFMT_MU_LAW + ? dmasound_ulaw2dma16 : dmasound_alaw2dma16); + unsigned int data = expand_data; + unsigned int *p = (unsigned int *) &frame[*frameUsed]; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + int stereo = dmasound.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 pmac_ctx_s8(const u_char __user *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 = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int stereo = dmasound.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 pmac_ctx_u8(const u_char __user *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 = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int stereo = dmasound.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 pmac_ctx_s16(const u_char __user *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 __user *up = (unsigned short __user *) userPtr; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int stereo = dmasound.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 pmac_ctx_u16(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + int mask = (dmasound.soft.format == AFMT_U16_LE? 0x0080: 0x8000); + unsigned int *p = (unsigned int *) &frame[*frameUsed]; + unsigned int data = expand_data; + unsigned short __user *up = (unsigned short __user *) userPtr; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int stereo = dmasound.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; +} + +/* data in routines... */ + +static ssize_t pmac_ct_s8_read(const u_char __user *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 = dmasound.soft.stereo; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + used = count = min_t(unsigned long, userCount, frameLeft); + while (count > 0) { + u_char data; + + val = *p++; + val = (val * software_input_volume) >> 7; + data = val >> 8; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + if (stereo) { + val = *p; + val = (val * software_input_volume) >> 7; + data = val >> 8; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + } + p++; + count--; + } + *frameUsed += used * 4; + return stereo? used * 2: used; +} + + +static ssize_t pmac_ct_u8_read(const u_char __user *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 = dmasound.soft.stereo; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + used = count = min_t(unsigned long, userCount, frameLeft); + while (count > 0) { + u_char data; + + val = *p++; + val = (val * software_input_volume) >> 7; + data = (val >> 8) ^ 0x80; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + if (stereo) { + val = *p; + val = (val * software_input_volume) >> 7; + data = (val >> 8) ^ 0x80; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + } + p++; + count--; + } + *frameUsed += used * 4; + return stereo? used * 2: used; +} + +static ssize_t pmac_ct_s16_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + int stereo = dmasound.soft.stereo; + short *fp = (short *) &frame[*frameUsed]; + short __user *up = (short __user *) userPtr; + + frameLeft >>= 2; + userCount >>= (stereo? 2: 1); + used = count = min_t(unsigned long, userCount, frameLeft); + while (count > 0) { + short data; + + data = *fp++; + data = (data * software_input_volume) >> 7; + if (put_user(data, up++)) + return -EFAULT; + if (stereo) { + data = *fp; + data = (data * software_input_volume) >> 7; + if (put_user(data, up++)) + return -EFAULT; + } + fp++; + count--; + } + *frameUsed += used * 4; + return stereo? used * 4: used * 2; +} + +static ssize_t pmac_ct_u16_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + int mask = (dmasound.soft.format == AFMT_U16_LE? 0x0080: 0x8000); + int stereo = dmasound.soft.stereo; + short *fp = (short *) &frame[*frameUsed]; + short __user *up = (short __user *) userPtr; + + frameLeft >>= 2; + userCount >>= (stereo? 2: 1); + used = count = min_t(unsigned long, userCount, frameLeft); + while (count > 0) { + int data; + + data = *fp++; + data = (data * software_input_volume) >> 7; + data ^= mask; + if (put_user(data, up++)) + return -EFAULT; + if (stereo) { + data = *fp; + data = (data * software_input_volume) >> 7; + data ^= mask; + if (put_user(data, up++)) + return -EFAULT; + } + fp++; + count--; + } + *frameUsed += used * 4; + return stereo? used * 4: used * 2; +} + +/* data in routines (reducing speed)... */ + +static ssize_t pmac_ctx_s8_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + short *p = (short *) &frame[*frameUsed]; + int bal = expand_read_bal; + int vall,valr, stereo = dmasound.soft.stereo; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char data; + + if (bal<0 && userCount == 0) + break; + vall = *p++; + vall = (vall * software_input_volume) >> 7; + if (stereo) { + valr = *p; + valr = (valr * software_input_volume) >> 7; + } + p++; + if (bal < 0) { + data = vall >> 8; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + if (stereo) { + data = valr >> 8; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + } + userCount--; + bal += hSpeed; + } + frameLeft--; + bal -= sSpeed; + } + expand_read_bal=bal; + *frameUsed += (ftotal - frameLeft) * 4; + utotal -= userCount; + return stereo? utotal * 2: utotal; +} + + +static ssize_t pmac_ctx_u8_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + short *p = (short *) &frame[*frameUsed]; + int bal = expand_read_bal; + int vall,valr, stereo = dmasound.soft.stereo; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + frameLeft >>= 2; + if (stereo) + userCount >>= 1; + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char data; + + if (bal<0 && userCount == 0) + break; + + vall = *p++; + vall = (vall * software_input_volume) >> 7; + if (stereo) { + valr = *p; + valr = (valr * software_input_volume) >> 7; + } + p++; + if (bal < 0) { + data = (vall >> 8) ^ 0x80; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + if (stereo) { + data = (valr >> 8) ^ 0x80; + if (put_user(data, (u_char __user *)userPtr++)) + return -EFAULT; + } + userCount--; + bal += hSpeed; + } + frameLeft--; + bal -= sSpeed; + } + expand_read_bal=bal; + *frameUsed += (ftotal - frameLeft) * 4; + utotal -= userCount; + return stereo? utotal * 2: utotal; +} + +static ssize_t pmac_ctx_s16_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + int bal = expand_read_bal; + short *fp = (short *) &frame[*frameUsed]; + short __user *up = (short __user *) userPtr; + int stereo = dmasound.soft.stereo; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + frameLeft >>= 2; + userCount >>= (stereo? 2: 1); + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + int datal,datar; + + if (bal<0 && userCount == 0) + break; + + datal = *fp++; + datal = (datal * software_input_volume) >> 7; + if (stereo) { + datar = *fp; + datar = (datar * software_input_volume) >> 7; + } + fp++; + if (bal < 0) { + if (put_user(datal, up++)) + return -EFAULT; + if (stereo) { + if (put_user(datar, up++)) + return -EFAULT; + } + userCount--; + bal += hSpeed; + } + frameLeft--; + bal -= sSpeed; + } + expand_read_bal=bal; + *frameUsed += (ftotal - frameLeft) * 4; + utotal -= userCount; + return stereo? utotal * 4: utotal * 2; +} + +static ssize_t pmac_ctx_u16_read(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + int bal = expand_read_bal; + int mask = (dmasound.soft.format == AFMT_U16_LE? 0x0080: 0x8000); + short *fp = (short *) &frame[*frameUsed]; + short __user *up = (short __user *) userPtr; + int stereo = dmasound.soft.stereo; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + frameLeft >>= 2; + userCount >>= (stereo? 2: 1); + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + int datal,datar; + + if (bal<0 && userCount == 0) + break; + + datal = *fp++; + datal = (datal * software_input_volume) >> 7; + datal ^= mask; + if (stereo) { + datar = *fp; + datar = (datar * software_input_volume) >> 7; + datar ^= mask; + } + fp++; + if (bal < 0) { + if (put_user(datal, up++)) + return -EFAULT; + if (stereo) { + if (put_user(datar, up++)) + return -EFAULT; + } + userCount--; + bal += hSpeed; + } + frameLeft--; + bal -= sSpeed; + } + expand_read_bal=bal; + *frameUsed += (ftotal - frameLeft) * 4; + utotal -= userCount; + return stereo? utotal * 4: utotal * 2; +} + + +TRANS transAwacsNormal = { + .ct_ulaw= pmac_ct_law, + .ct_alaw= pmac_ct_law, + .ct_s8= pmac_ct_s8, + .ct_u8= pmac_ct_u8, + .ct_s16be= pmac_ct_s16, + .ct_u16be= pmac_ct_u16, + .ct_s16le= pmac_ct_s16, + .ct_u16le= pmac_ct_u16, +}; + +TRANS transAwacsExpand = { + .ct_ulaw= pmac_ctx_law, + .ct_alaw= pmac_ctx_law, + .ct_s8= pmac_ctx_s8, + .ct_u8= pmac_ctx_u8, + .ct_s16be= pmac_ctx_s16, + .ct_u16be= pmac_ctx_u16, + .ct_s16le= pmac_ctx_s16, + .ct_u16le= pmac_ctx_u16, +}; + +TRANS transAwacsNormalRead = { + .ct_s8= pmac_ct_s8_read, + .ct_u8= pmac_ct_u8_read, + .ct_s16be= pmac_ct_s16_read, + .ct_u16be= pmac_ct_u16_read, + .ct_s16le= pmac_ct_s16_read, + .ct_u16le= pmac_ct_u16_read, +}; + +TRANS transAwacsExpandRead = { + .ct_s8= pmac_ctx_s8_read, + .ct_u8= pmac_ctx_u8_read, + .ct_s16be= pmac_ctx_s16_read, + .ct_u16be= pmac_ctx_u16_read, + .ct_s16le= pmac_ctx_s16_read, + .ct_u16le= pmac_ctx_u16_read, +}; + +/* translation tables */ +/* 16 bit mu-law */ + +static short dmasound_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 dmasound_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, +}; |