/* * 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 <asm/uaccess.h> #include <asm/setup.h> #include <asm/amigahw.h> #include <asm/amigaints.h> #include <asm/machdep.h> #include "dmasound.h" #define DMASOUND_PAULA_REVISION 0 #define DMASOUND_PAULA_EDITION 4 #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", .owner = THIS_MODULE, }, }; static int __init amiga_audio_init(void) { return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe); } module_init(amiga_audio_init); static void __exit amiga_audio_exit(void) { platform_driver_unregister(&amiga_audio_driver); } module_exit(amiga_audio_exit); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:amiga-audio");