Linux-2.6.12-rc2
Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.
Let it rip!
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,
+};