739 lines
19 KiB
C
739 lines
19 KiB
C
/*
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* linux/sound/oss/dmasound/dmasound_paula.c
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*
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* Amiga `Paula' DMA Sound Driver
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*
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* See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
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* prior to 28/01/2001
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*
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* 28/01/2001 [0.1] Iain Sandoe
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* - added versioning
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* - put in and populated the hardware_afmts field.
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* [0.2] - put in SNDCTL_DSP_GETCAPS value.
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* [0.3] - put in constraint on state buffer usage.
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* [0.4] - put in default hard/soft settings
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*/
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#include <linux/module.h>
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#include <linux/mm.h>
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#include <linux/init.h>
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#include <linux/ioport.h>
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#include <linux/soundcard.h>
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#include <linux/interrupt.h>
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#include <linux/platform_device.h>
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#include <linux/uaccess.h>
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#include <asm/setup.h>
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#include <asm/amigahw.h>
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#include <asm/amigaints.h>
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#include <asm/machdep.h>
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#include "dmasound.h"
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#define DMASOUND_PAULA_REVISION 0
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#define DMASOUND_PAULA_EDITION 4
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#define custom amiga_custom
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/*
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* The minimum period for audio depends on htotal (for OCS/ECS/AGA)
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* (Imported from arch/m68k/amiga/amisound.c)
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*/
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extern volatile u_short amiga_audio_min_period;
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/*
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* amiga_mksound() should be able to restore the period after beeping
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* (Imported from arch/m68k/amiga/amisound.c)
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*/
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extern u_short amiga_audio_period;
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/*
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* Audio DMA masks
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*/
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#define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
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#define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
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#define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
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/*
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* Helper pointers for 16(14)-bit sound
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*/
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static int write_sq_block_size_half, write_sq_block_size_quarter;
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/*** Low level stuff *********************************************************/
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static void *AmiAlloc(unsigned int size, gfp_t flags);
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static void AmiFree(void *obj, unsigned int size);
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static int AmiIrqInit(void);
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#ifdef MODULE
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static void AmiIrqCleanUp(void);
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#endif
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static void AmiSilence(void);
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static void AmiInit(void);
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static int AmiSetFormat(int format);
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static int AmiSetVolume(int volume);
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static int AmiSetTreble(int treble);
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static void AmiPlayNextFrame(int index);
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static void AmiPlay(void);
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static irqreturn_t AmiInterrupt(int irq, void *dummy);
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#ifdef CONFIG_HEARTBEAT
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/*
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* Heartbeat interferes with sound since the 7 kHz low-pass filter and the
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* power LED are controlled by the same line.
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*/
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static void (*saved_heartbeat)(int) = NULL;
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static inline void disable_heartbeat(void)
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{
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if (mach_heartbeat) {
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saved_heartbeat = mach_heartbeat;
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mach_heartbeat = NULL;
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}
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AmiSetTreble(dmasound.treble);
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}
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static inline void enable_heartbeat(void)
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{
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if (saved_heartbeat)
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mach_heartbeat = saved_heartbeat;
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}
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#else /* !CONFIG_HEARTBEAT */
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#define disable_heartbeat() do { } while (0)
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#define enable_heartbeat() do { } while (0)
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#endif /* !CONFIG_HEARTBEAT */
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/*** Mid level stuff *********************************************************/
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static void AmiMixerInit(void);
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static int AmiMixerIoctl(u_int cmd, u_long arg);
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static int AmiWriteSqSetup(void);
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static int AmiStateInfo(char *buffer, size_t space);
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/*** Translations ************************************************************/
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/* ++TeSche: radically changed for new expanding purposes...
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*
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* These two routines now deal with copying/expanding/translating the samples
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* from user space into our buffer at the right frequency. They take care about
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* how much data there's actually to read, how much buffer space there is and
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* to convert samples into the right frequency/encoding. They will only work on
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* complete samples so it may happen they leave some bytes in the input stream
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* if the user didn't write a multiple of the current sample size. They both
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* return the number of bytes they've used from both streams so you may detect
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* such a situation. Luckily all programs should be able to cope with that.
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*
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* I think I've optimized anything as far as one can do in plain C, all
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* variables should fit in registers and the loops are really short. There's
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* one loop for every possible situation. Writing a more generalized and thus
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* parameterized loop would only produce slower code. Feel free to optimize
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* this in assembler if you like. :)
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*
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* I think these routines belong here because they're not yet really hardware
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* independent, especially the fact that the Falcon can play 16bit samples
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* only in stereo is hardcoded in both of them!
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*
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* ++geert: split in even more functions (one per format)
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*/
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/*
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* Native format
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*/
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static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
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u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
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{
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ssize_t count, used;
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if (!dmasound.soft.stereo) {
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void *p = &frame[*frameUsed];
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count = min_t(unsigned long, userCount, frameLeft) & ~1;
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used = count;
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if (copy_from_user(p, userPtr, count))
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return -EFAULT;
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} else {
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u_char *left = &frame[*frameUsed>>1];
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u_char *right = left+write_sq_block_size_half;
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count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
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used = count*2;
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while (count > 0) {
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if (get_user(*left++, userPtr++)
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|| get_user(*right++, userPtr++))
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return -EFAULT;
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count--;
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}
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}
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*frameUsed += used;
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return used;
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}
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/*
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* Copy and convert 8 bit data
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*/
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#define GENERATE_AMI_CT8(funcname, convsample) \
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static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
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u_char frame[], ssize_t *frameUsed, \
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ssize_t frameLeft) \
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{ \
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ssize_t count, used; \
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\
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if (!dmasound.soft.stereo) { \
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u_char *p = &frame[*frameUsed]; \
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count = min_t(size_t, userCount, frameLeft) & ~1; \
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used = count; \
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while (count > 0) { \
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u_char data; \
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if (get_user(data, userPtr++)) \
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return -EFAULT; \
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*p++ = convsample(data); \
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count--; \
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} \
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} else { \
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u_char *left = &frame[*frameUsed>>1]; \
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u_char *right = left+write_sq_block_size_half; \
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count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
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used = count*2; \
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while (count > 0) { \
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u_char data; \
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if (get_user(data, userPtr++)) \
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return -EFAULT; \
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*left++ = convsample(data); \
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if (get_user(data, userPtr++)) \
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return -EFAULT; \
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*right++ = convsample(data); \
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count--; \
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} \
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} \
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*frameUsed += used; \
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return used; \
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}
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#define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)])
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#define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)])
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#define AMI_CT_U8(x) ((x) ^ 0x80)
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GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
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GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
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GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
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/*
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* Copy and convert 16 bit data
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*/
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#define GENERATE_AMI_CT_16(funcname, convsample) \
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static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
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u_char frame[], ssize_t *frameUsed, \
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ssize_t frameLeft) \
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{ \
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const u_short __user *ptr = (const u_short __user *)userPtr; \
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ssize_t count, used; \
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u_short data; \
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\
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if (!dmasound.soft.stereo) { \
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u_char *high = &frame[*frameUsed>>1]; \
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u_char *low = high+write_sq_block_size_half; \
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count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
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used = count*2; \
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while (count > 0) { \
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if (get_user(data, ptr++)) \
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return -EFAULT; \
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data = convsample(data); \
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*high++ = data>>8; \
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*low++ = (data>>2) & 0x3f; \
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count--; \
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} \
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} else { \
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u_char *lefth = &frame[*frameUsed>>2]; \
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u_char *leftl = lefth+write_sq_block_size_quarter; \
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u_char *righth = lefth+write_sq_block_size_half; \
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u_char *rightl = righth+write_sq_block_size_quarter; \
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count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \
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used = count*4; \
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while (count > 0) { \
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if (get_user(data, ptr++)) \
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return -EFAULT; \
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data = convsample(data); \
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*lefth++ = data>>8; \
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*leftl++ = (data>>2) & 0x3f; \
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if (get_user(data, ptr++)) \
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return -EFAULT; \
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data = convsample(data); \
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*righth++ = data>>8; \
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*rightl++ = (data>>2) & 0x3f; \
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count--; \
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} \
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} \
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*frameUsed += used; \
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return used; \
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}
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#define AMI_CT_S16BE(x) (x)
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#define AMI_CT_U16BE(x) ((x) ^ 0x8000)
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#define AMI_CT_S16LE(x) (le2be16((x)))
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#define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000)
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GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
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GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
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GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
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GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
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static TRANS transAmiga = {
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.ct_ulaw = ami_ct_ulaw,
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.ct_alaw = ami_ct_alaw,
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.ct_s8 = ami_ct_s8,
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.ct_u8 = ami_ct_u8,
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.ct_s16be = ami_ct_s16be,
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.ct_u16be = ami_ct_u16be,
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.ct_s16le = ami_ct_s16le,
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.ct_u16le = ami_ct_u16le,
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};
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/*** Low level stuff *********************************************************/
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static inline void StopDMA(void)
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{
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custom.aud[0].audvol = custom.aud[1].audvol = 0;
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custom.aud[2].audvol = custom.aud[3].audvol = 0;
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custom.dmacon = AMI_AUDIO_OFF;
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enable_heartbeat();
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}
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static void *AmiAlloc(unsigned int size, gfp_t flags)
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{
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return amiga_chip_alloc((long)size, "dmasound [Paula]");
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}
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static void AmiFree(void *obj, unsigned int size)
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{
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amiga_chip_free (obj);
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}
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static int __init AmiIrqInit(void)
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{
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/* turn off DMA for audio channels */
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StopDMA();
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/* Register interrupt handler. */
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if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
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AmiInterrupt))
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return 0;
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return 1;
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}
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#ifdef MODULE
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static void AmiIrqCleanUp(void)
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{
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/* turn off DMA for audio channels */
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StopDMA();
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/* release the interrupt */
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free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
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}
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#endif /* MODULE */
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static void AmiSilence(void)
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{
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/* turn off DMA for audio channels */
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StopDMA();
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}
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static void AmiInit(void)
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{
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int period, i;
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AmiSilence();
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if (dmasound.soft.speed)
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period = amiga_colorclock/dmasound.soft.speed-1;
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else
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period = amiga_audio_min_period;
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dmasound.hard = dmasound.soft;
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dmasound.trans_write = &transAmiga;
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if (period < amiga_audio_min_period) {
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/* we would need to squeeze the sound, but we won't do that */
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period = amiga_audio_min_period;
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} else if (period > 65535) {
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period = 65535;
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}
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dmasound.hard.speed = amiga_colorclock/(period+1);
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for (i = 0; i < 4; i++)
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custom.aud[i].audper = period;
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amiga_audio_period = period;
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}
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static int AmiSetFormat(int format)
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{
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int size;
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/* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
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switch (format) {
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case AFMT_QUERY:
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return dmasound.soft.format;
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case AFMT_MU_LAW:
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case AFMT_A_LAW:
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case AFMT_U8:
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case AFMT_S8:
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size = 8;
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break;
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case AFMT_S16_BE:
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case AFMT_U16_BE:
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case AFMT_S16_LE:
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case AFMT_U16_LE:
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size = 16;
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break;
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default: /* :-) */
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size = 8;
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format = AFMT_S8;
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}
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dmasound.soft.format = format;
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dmasound.soft.size = size;
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if (dmasound.minDev == SND_DEV_DSP) {
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dmasound.dsp.format = format;
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dmasound.dsp.size = dmasound.soft.size;
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}
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AmiInit();
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return format;
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}
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#define VOLUME_VOXWARE_TO_AMI(v) \
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(((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
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#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
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static int AmiSetVolume(int volume)
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{
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dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
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custom.aud[0].audvol = dmasound.volume_left;
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dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
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custom.aud[1].audvol = dmasound.volume_right;
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if (dmasound.hard.size == 16) {
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if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
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custom.aud[2].audvol = 1;
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custom.aud[3].audvol = 1;
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} else {
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custom.aud[2].audvol = 0;
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custom.aud[3].audvol = 0;
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}
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}
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return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
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(VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
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}
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static int AmiSetTreble(int treble)
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{
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dmasound.treble = treble;
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if (treble < 50)
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ciaa.pra &= ~0x02;
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else
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ciaa.pra |= 0x02;
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return treble;
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}
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#define AMI_PLAY_LOADED 1
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#define AMI_PLAY_PLAYING 2
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#define AMI_PLAY_MASK 3
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static void AmiPlayNextFrame(int index)
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{
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u_char *start, *ch0, *ch1, *ch2, *ch3;
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u_long size;
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/* used by AmiPlay() if all doubts whether there really is something
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* to be played are already wiped out.
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*/
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start = write_sq.buffers[write_sq.front];
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size = (write_sq.count == index ? write_sq.rear_size
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: write_sq.block_size)>>1;
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if (dmasound.hard.stereo) {
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ch0 = start;
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ch1 = start+write_sq_block_size_half;
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size >>= 1;
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} else {
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ch0 = start;
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ch1 = start;
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}
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disable_heartbeat();
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custom.aud[0].audvol = dmasound.volume_left;
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custom.aud[1].audvol = dmasound.volume_right;
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if (dmasound.hard.size == 8) {
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custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
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custom.aud[0].audlen = size;
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custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
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custom.aud[1].audlen = size;
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custom.dmacon = AMI_AUDIO_8;
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} else {
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size >>= 1;
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custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
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custom.aud[0].audlen = size;
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custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
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custom.aud[1].audlen = size;
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if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
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/* We can play pseudo 14-bit only with the maximum volume */
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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");
|