projects / linux.git / blame
| Commit | Line | Data |
|---|---|---|
| 1da177e4 LT |
1 | /* |
| 2 | * pas2_pcm.c Audio routines for PAS16 | |
| 3 | * | |
| 4 | * | |
| 5 | * Copyright (C) by Hannu Savolainen 1993-1997 | |
| 6 | * | |
| 7 | * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) | |
| 8 | * Version 2 (June 1991). See the "COPYING" file distributed with this software | |
| 9 | * for more info. | |
| 10 | * | |
| 11 | * | |
| 12 | * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) | |
| 13 | * Alan Cox : Swatted a double allocation of device bug. Made a few | |
| 14 | * more things module options. | |
| 15 | * Bartlomiej Zolnierkiewicz : Added __init to pas_pcm_init() | |
| 16 | */ | |
| 17 | ||
| 18 | #include <linux/init.h> | |
| 19 | #include <linux/spinlock.h> | |
| 20 | #include <asm/timex.h> | |
| 21 | #include "sound_config.h" | |
| 22 | ||
| 23 | #include "pas2.h" | |
| 24 | ||
| 25 | #ifndef DEB | |
| 26 | #define DEB(WHAT) | |
| 27 | #endif | |
| 28 | ||
| 29 | #define PAS_PCM_INTRBITS (0x08) | |
| 30 | /* | |
| 31 | * Sample buffer timer interrupt enable | |
| 32 | */ | |
| 33 | ||
| 34 | #define PCM_NON 0 | |
| 35 | #define PCM_DAC 1 | |
| 36 | #define PCM_ADC 2 | |
| 37 | ||
| 38 | static unsigned long pcm_speed; /* sampling rate */ | |
| 39 | static unsigned char pcm_channels = 1; /* channels (1 or 2) */ | |
| 40 | static unsigned char pcm_bits = 8; /* bits/sample (8 or 16) */ | |
| 41 | static unsigned char pcm_filter; /* filter FLAG */ | |
| 42 | static unsigned char pcm_mode = PCM_NON; | |
| 43 | static unsigned long pcm_count; | |
| 44 | static unsigned short pcm_bitsok = 8; /* mask of OK bits */ | |
| 45 | static int pcm_busy; | |
| 46 | int pas_audiodev = -1; | |
| 47 | static int open_mode; | |
| 48 | ||
| 49 | extern spinlock_t pas_lock; | |
| 50 | ||
| 51 | static int pcm_set_speed(int arg) | |
| 52 | { | |
| 53 | int foo, tmp; | |
| 54 | unsigned long flags; | |
| 55 | ||
| 56 | if (arg == 0) | |
| 57 | return pcm_speed; | |
| 58 | ||
| 59 | if (arg > 44100) | |
| 60 | arg = 44100; | |
| 61 | if (arg < 5000) | |
| 62 | arg = 5000; | |
| 63 | ||
| 64 | if (pcm_channels & 2) | |
| 65 | { | |
| 66 | foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg; | |
| 67 | arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo; | |
| 68 | } | |
| 69 | else | |
| 70 | { | |
| 71 | foo = (CLOCK_TICK_RATE + (arg / 2)) / arg; | |
| 72 | arg = (CLOCK_TICK_RATE + (foo / 2)) / foo; | |
| 73 | } | |
| 74 | ||
| 75 | pcm_speed = arg; | |
| 76 | ||
| 77 | tmp = pas_read(0x0B8A); | |
| 78 | ||
| 79 | /* | |
| 80 | * Set anti-aliasing filters according to sample rate. You really *NEED* | |
| 81 | * to enable this feature for all normal recording unless you want to | |
| 82 | * experiment with aliasing effects. | |
| 83 | * These filters apply to the selected "recording" source. | |
| 84 | * I (pfw) don't know the encoding of these 5 bits. The values shown | |
| 85 | * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/. | |
| 86 | * | |
| 87 | * I cleared bit 5 of these values, since that bit controls the master | |
| 88 |