diff options
Diffstat (limited to 'ANDROID_3.4.5/sound/oss/dmasound/dmasound_paula.c')
| -rw-r--r-- | ANDROID_3.4.5/sound/oss/dmasound/dmasound_paula.c | 751 |
1 files changed, 0 insertions, 751 deletions
diff --git a/ANDROID_3.4.5/sound/oss/dmasound/dmasound_paula.c b/ANDROID_3.4.5/sound/oss/dmasound/dmasound_paula.c deleted file mode 100644 index 87910e99..00000000 --- a/ANDROID_3.4.5/sound/oss/dmasound/dmasound_paula.c +++ /dev/null @@ -1,751 +0,0 @@ -/* - * 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"); |