[Feature] add GA346 baseline version
Change-Id: Ic62933698569507dcf98240cdf5d9931ae34348f
diff --git a/src/kernel/linux/v4.19/sound/oss/.gitignore b/src/kernel/linux/v4.19/sound/oss/.gitignore
new file mode 100644
index 0000000..12a3920
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/.gitignore
@@ -0,0 +1,3 @@
+#Ignore generated files
+pss_boot.h
+trix_boot.h
diff --git a/src/kernel/linux/v4.19/sound/oss/dmasound/Kconfig b/src/kernel/linux/v4.19/sound/oss/dmasound/Kconfig
new file mode 100644
index 0000000..f456574
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/Kconfig
@@ -0,0 +1,45 @@
+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_PAULA
+ tristate "Amiga DMA sound support"
+ depends on AMIGA && 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
+ select SOUND_OSS_CORE
diff --git a/src/kernel/linux/v4.19/sound/oss/dmasound/Makefile b/src/kernel/linux/v4.19/sound/oss/dmasound/Makefile
new file mode 100644
index 0000000..3c15316
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/Makefile
@@ -0,0 +1,7 @@
+#
+# Makefile for the DMA sound driver
+#
+
+obj-$(CONFIG_DMASOUND_ATARI) += dmasound_core.o dmasound_atari.o
+obj-$(CONFIG_DMASOUND_PAULA) += dmasound_core.o dmasound_paula.o
+obj-$(CONFIG_DMASOUND_Q40) += dmasound_core.o dmasound_q40.o
diff --git a/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound.h b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound.h
new file mode 100644
index 0000000..c1c52b4
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound.h
@@ -0,0 +1,262 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#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>
+
+#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
+
+#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)
+
+ /*
+ * 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, gfp_t);
+ void (*dma_free)(void *, unsigned int);
+ int (*irqinit)(void);
+#ifdef MODULE
+ void (*irqcleanup)(void);
+#endif
+ void (*init)(void);
+ void (*silence)(void);
+ int (*setFormat)(int);
+ int (*setVolume)(int);
+ int (*setBass)(int);
+ int (*setTreble)(int);
+ int (*setGain)(int);
+ void (*play)(void);
+ void (*record)(void); /* optional */
+ void (*mixer_init)(void); /* optional */
+ int (*mixer_ioctl)(u_int, u_long); /* optional */
+ int (*write_sq_setup)(void); /* optional */
+ int (*read_sq_setup)(void); /* optional */
+ int (*sq_open)(fmode_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 */
+ 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 non_blocking;
+ int busy, syncing, xruns, died;
+};
+
+#define WAKE_UP(queue) (wake_up_interruptible(&queue))
+
+extern struct sound_queue dmasound_write_sq;
+#define write_sq dmasound_write_sq
+
+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_read_bal; /* Balance factor for reading */
+extern uint software_input_volume; /* software implemented recording volume! */
+
+#endif /* _dmasound_h_ */
diff --git a/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_atari.c b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_atari.c
new file mode 100644
index 0000000..a1a2979
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_atari.c
@@ -0,0 +1,1620 @@
+/*
+ * 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 <linux/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 __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_s16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_u16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_s16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_u16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_s16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_u16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_s16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_u16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *AtaAlloc(unsigned int size, gfp_t 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);
+
+/*** 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(fmode_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 __user *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 __user *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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ *p++ = data ^ 0x8080;
+ count--;
+ }
+ }
+ *frameUsed += used;
+ return used;
+}
+
+
+static ssize_t ata_ct_s16be(const u_char __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ *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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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_int data;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ *p++ = data ^ 0x80008000;
+ count--;
+ }
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ct_s16le(const u_char __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data = le2be16dbl(data);
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ct_u16le(const u_char __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data = le2be16dbl(data) ^ 0x80008000;
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ctx_law(const u_char __user *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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ 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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ 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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ 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 __user *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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ 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 __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ 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, gfp_t 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. */
+ st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */
+ st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */
+ st_mfp.tim_ct_a = 8; /* Turn on event counting. */
+ /* Register interrupt handler. */
+ if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, 0, "DMA sound",
+ AtaInterrupt))
+ return 0;
+ st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */
+ st_mfp.int_mk_a |= 0x20;
+ return 1;
+}
+
+#ifdef MODULE
+static void AtaIrqCleanUp(void)
+{
+ st_mfp.tim_ct_a = 0; /* stop timer */
+ st_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)
+{
+#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;
+ goto out;
+ }
+
+ 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);
+ goto out;
+ }
+
+ /* 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. */
+out:
+ 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(fmode_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 __initdata 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 __initdata 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 ((st_mfp.int_en_a & st_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/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_core.c b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_core.c
new file mode 100644
index 0000000..fc9bcd4
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_core.c
@@ -0,0 +1,1600 @@
+/*
+ * 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/mutex.h>
+#include <linux/sched/signal.h>
+
+#include <linux/uaccess.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_CORE_REVISION 1
+#define DMASOUND_CORE_EDITION 6
+
+ /*
+ * Declarations
+ */
+
+static DEFINE_MUTEX(dmasound_core_mutex);
+int dmasound_catchRadius = 0;
+module_param(dmasound_catchRadius, int, 0);
+
+static unsigned int numWriteBufs = DEFAULT_N_BUFFERS;
+module_param(numWriteBufs, int, 0);
+static unsigned int writeBufSize = DEFAULT_BUFF_SIZE ; /* in bytes */
+module_param(writeBufSize, int, 0);
+
+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 */
+
+/* control over who can modify resources shared between play/record */
+static fmode_t shared_resource_owner;
+static int shared_resources_initialised;
+
+ /*
+ * Mid level stuff
+ */
+
+struct sound_settings dmasound = {
+ .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock)
+};
+
+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)
+{
+ mutex_lock(&dmasound_core_mutex);
+ if (!try_module_get(dmasound.mach.owner)) {
+ mutex_unlock(&dmasound_core_mutex);
+ return -ENODEV;
+ }
+ mixer.busy = 1;
+ mutex_unlock(&dmasound_core_mutex);
+ return 0;
+}
+
+static int mixer_release(struct inode *inode, struct file *file)
+{
+ mutex_lock(&dmasound_core_mutex);
+ mixer.busy = 0;
+ module_put(dmasound.mach.owner);
+ mutex_unlock(&dmasound_core_mutex);
+ return 0;
+}
+
+static int mixer_ioctl(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 long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg)
+{
+ int ret;
+
+ mutex_lock(&dmasound_core_mutex);
+ ret = mixer_ioctl(file, cmd, arg);
+ mutex_unlock(&dmasound_core_mutex);
+
+ return ret;
+}
+
+static const struct file_operations mixer_fops =
+{
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .unlocked_ioctl = mixer_unlocked_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) ;
+
+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_array (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;
+ }
+ 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) {
+ DEFINE_WAIT(wait);
+
+ while (write_sq.count >= write_sq.max_active) {
+ prepare_to_wait(&write_sq.action_queue, &wait, TASK_INTERRUPTIBLE);
+ sq_play();
+ if (write_sq.non_blocking) {
+ finish_wait(&write_sq.action_queue, &wait);
+ return uWritten > 0 ? uWritten : -EAGAIN;
+ }
+ if (write_sq.count < write_sq.max_active)
+ break;
+
+ schedule_timeout(HZ);
+ if (signal_pending(current)) {
+ finish_wait(&write_sq.action_queue, &wait);
+ return uWritten > 0 ? uWritten : -EINTR;
+ }
+ }
+
+ finish_wait(&write_sq.action_queue, &wait);
+
+ /* 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 __poll_t sq_poll(struct file *file, struct poll_table_struct *wait)
+{
+ __poll_t 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);
+ if (file->f_mode & FMODE_WRITE)
+ if (write_sq.count < write_sq.max_active || write_sq.block_size - write_sq.rear_size > 0)
+ mask |= EPOLLOUT | EPOLLWRNORM;
+ return mask;
+
+}
+
+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,
+ fmode_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, fmode_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;
+ if (wait_event_interruptible(sq->open_queue, !sq->busy))
+ 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->non_blocking = file->f_flags & O_NONBLOCK;
+ }
+ 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 )
+
+static int sq_open(struct inode *inode, struct file *file)
+{
+ int rc;
+
+ mutex_lock(&dmasound_core_mutex);
+ if (!try_module_get(dmasound.mach.owner)) {
+ mutex_unlock(&dmasound_core_mutex);
+ return -ENODEV;
+ }
+
+ rc = write_sq_open(file); /* checks the f_mode */
+ if (rc)
+ goto out;
+ 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;
+ }
+
+ 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
+ mutex_unlock(&dmasound_core_mutex);
+ return 0;
+ out:
+ module_put(dmasound.mach.owner);
+ mutex_unlock(&dmasound_core_mutex);
+ 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 ;
+}
+
+static void sq_reset(void)
+{
+ sq_reset_output() ;
+ /* 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(void)
+{
+ int rc = 0;
+ int timeout = 5;
+
+ write_sq.syncing |= 1;
+ sq_play(); /* there may be an incomplete frame waiting */
+
+ while (write_sq.active) {
+ wait_event_interruptible_timeout(write_sq.sync_queue,
+ !write_sq.active, HZ);
+ 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;
+
+ mutex_lock(&dmasound_core_mutex);
+
+ if (file->f_mode & FMODE_WRITE) {
+ if (write_sq.busy)
+ rc = sq_fsync();
+
+ 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() */
+
+ mutex_unlock(&dmasound_core_mutex);
+
+ 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(fmode_t md)
+{
+ if (shared_resource_owner)
+ return (shared_resource_owner & md) != 0;
+ 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)
+{
+ 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 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 ;
+ 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 immediately.
+ */
+ result = 0 ;
+ if (file->f_mode & FMODE_WRITE) {
+ result = sq_fsync();
+ 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;
+ 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(file, cmd, arg);
+ }
+ return -EINVAL;
+}
+
+static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg)
+{
+ int ret;
+
+ mutex_lock(&dmasound_core_mutex);
+ ret = sq_ioctl(file, cmd, arg);
+ mutex_unlock(&dmasound_core_mutex);
+
+ return ret;
+}
+
+static const struct file_operations sq_fops =
+{
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .write = sq_write,
+ .poll = sq_poll,
+ .unlocked_ioctl = sq_unlocked_ioctl,
+ .open = sq_open,
+ .release = sq_release,
+};
+
+static int sq_init(void)
+{
+ const struct file_operations *fops = &sq_fops;
+#ifndef MODULE
+ int sq_unit;
+#endif
+
+ sq_unit = register_sound_dsp(fops, -1);
+ if (sq_unit < 0) {
+ printk(KERN_ERR "dmasound_core: couldn't register fops\n") ;
+ return sq_unit ;
+ }
+
+ write_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.
+*/
+
+#define STAT_BUFF_LEN 768
+
+/* 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 */
+
+static 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;
+ int ret;
+
+ mutex_lock(&dmasound_core_mutex);
+ ret = -EBUSY;
+ if (state.busy)
+ goto out;
+
+ ret = -ENODEV;
+ if (!try_module_get(dmasound.mach.owner))
+ goto out;
+
+ 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) ;
+ 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 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;
+ ret = 0;
+out:
+ mutex_unlock(&dmasound_core_mutex);
+ return ret;
+}
+
+static int state_release(struct inode *inode, struct file *file)
+{
+ mutex_lock(&dmasound_core_mutex);
+ state.busy = 0;
+ module_put(dmasound.mach.owner);
+ mutex_unlock(&dmasound_core_mutex);
+ 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 const 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) ;
+ return 0;
+}
+
+#ifdef MODULE
+
+void dmasound_deinit(void)
+{
+ if (irq_installed) {
+ sound_silence();
+ dmasound.mach.irqcleanup();
+ irq_installed = 0;
+ }
+
+ write_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]) {
+ case 3:
+ if ((ints[3] < 0) || (ints[3] > MAX_CATCH_RADIUS))
+ printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius);
+ else
+ catchRadius = ints[3];
+ /* fall through */
+ case 2:
+ if (ints[1] < MIN_BUFFERS)
+ printk("dmasound_setup: invalid number of buffers, using default = %d\n", 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);
+EXPORT_SYMBOL(dmasound_catchRadius);
+#ifdef HAS_8BIT_TABLES
+EXPORT_SYMBOL(dmasound_ulaw2dma8);
+EXPORT_SYMBOL(dmasound_alaw2dma8);
+#endif
diff --git a/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_paula.c b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_paula.c
new file mode 100644
index 0000000..81eb82c
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_paula.c
@@ -0,0 +1,738 @@
+/*
+ * 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/mm.h>
+#include <linux/init.h>
+#include <linux/ioport.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <linux/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
+
+#define custom amiga_custom
+ /*
+ * 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, gfp_t 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);
+
+#ifdef CONFIG_HEARTBEAT
+
+ /*
+ * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
+ * power LED are controlled by the same line.
+ */
+
+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 __user *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 __user *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 __user *userPtr, size_t userCount, \
+ u_char frame[], ssize_t *frameUsed, \
+ ssize_t frameLeft) \
+{ \
+ const u_short __user *ptr = (const u_short __user *)userPtr; \
+ 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, ptr++)) \
+ 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, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *lefth++ = data>>8; \
+ *leftl++ = (data>>2) & 0x3f; \
+ if (get_user(data, ptr++)) \
+ 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, gfp_t 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)
+{
+ 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: overflowed 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 **********************************************************/
+
+
+static int __init amiga_audio_probe(struct platform_device *pdev)
+{
+ dmasound.mach = machAmiga;
+ dmasound.mach.default_hard = def_hard ;
+ dmasound.mach.default_soft = def_soft ;
+ return dmasound_init();
+}
+
+static int __exit amiga_audio_remove(struct platform_device *pdev)
+{
+ dmasound_deinit();
+ return 0;
+}
+
+static struct platform_driver amiga_audio_driver = {
+ .remove = __exit_p(amiga_audio_remove),
+ .driver = {
+ .name = "amiga-audio",
+ },
+};
+
+module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:amiga-audio");
diff --git a/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_q40.c b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_q40.c
new file mode 100644
index 0000000..be4fe15
--- /dev/null
+++ b/src/kernel/linux/v4.19/sound/oss/dmasound/dmasound_q40.c
@@ -0,0 +1,638 @@
+/*
+ * 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 <linux/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, gfp_t 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);
+static irqreturn_t Q40MonoInterrupt(int irq, void *dummy);
+static void Q40Interrupt(void);
+
+
+/*** Mid level stuff *********************************************************/
+
+
+
+/* userCount, frameUsed, frameLeft == byte counts */
+static ssize_t q40_ct_law(const u_char __user *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 __user *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 __user *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 __user *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 __user *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 __user *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 __user *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 __user *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 __user *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, gfp_t 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. */
+ if (request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0,
+ "DMA sound", Q40Interrupt))
+ return 0;
+
+ 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;
+ int error;
+
+ /* 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)
+ error = request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0,
+ "Q40 sound", Q40Interrupt);
+ else
+ error = request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0,
+ "Q40 sound", Q40Interrupt);
+ if (error && printk_ratelimit())
+ pr_err("Couldn't register sound interrupt\n");
+
+ 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)
+{
+ 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)
+{
+ 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 **********************************************************/
+
+
+static 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");