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Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
1330 lines
34 KiB
C
1330 lines
34 KiB
C
/*
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drivers/sound/harmony.c
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This is a sound driver for ASP's and Lasi's Harmony sound chip
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and is unlikely to be used for anything other than on a HP PA-RISC.
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Harmony is found in HP 712s, 715/new and many other GSC based machines.
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On older 715 machines you'll find the technically identical chip
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called 'Vivace'. Both Harmony and Vicace are supported by this driver.
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Copyright 2000 (c) Linuxcare Canada, Alex deVries <alex@onefishtwo.ca>
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Copyright 2000-2003 (c) Helge Deller <deller@gmx.de>
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Copyright 2001 (c) Matthieu Delahaye <delahaym@esiee.fr>
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Copyright 2001 (c) Jean-Christophe Vaugeois <vaugeoij@esiee.fr>
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Copyright 2004 (c) Stuart Brady <sdbrady@ntlworld.com>
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TODO:
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- fix SNDCTL_DSP_GETOSPACE and SNDCTL_DSP_GETISPACE ioctls to
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return the real values
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- add private ioctl for selecting line- or microphone input
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(only one of them is available at the same time)
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- add module parameters
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- implement mmap functionality
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- implement gain meter ?
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- ...
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*/
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#include <linux/delay.h>
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#include <linux/errno.h>
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#include <linux/init.h>
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#include <linux/interrupt.h>
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#include <linux/ioport.h>
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#include <linux/types.h>
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#include <linux/mm.h>
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#include <linux/pci.h>
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#include <asm/parisc-device.h>
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#include <asm/io.h>
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#include "sound_config.h"
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#define PFX "harmony: "
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#define HARMONY_VERSION "V0.9a"
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#undef DEBUG
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#ifdef DEBUG
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# define DPRINTK printk
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#else
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# define DPRINTK(x,...)
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#endif
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#define MAX_BUFS 10 /* maximum number of rotating buffers */
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#define HARMONY_BUF_SIZE 4096 /* needs to be a multiple of PAGE_SIZE (4096)! */
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#define CNTL_C 0x80000000
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#define CNTL_ST 0x00000020
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#define CNTL_44100 0x00000015 /* HARMONY_SR_44KHZ */
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#define CNTL_8000 0x00000008 /* HARMONY_SR_8KHZ */
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#define GAINCTL_HE 0x08000000
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#define GAINCTL_LE 0x04000000
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#define GAINCTL_SE 0x02000000
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#define DSTATUS_PN 0x00000200
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#define DSTATUS_RN 0x00000002
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#define DSTATUS_IE 0x80000000
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#define HARMONY_DF_16BIT_LINEAR 0
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#define HARMONY_DF_8BIT_ULAW 1
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#define HARMONY_DF_8BIT_ALAW 2
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#define HARMONY_SS_MONO 0
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#define HARMONY_SS_STEREO 1
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#define HARMONY_SR_8KHZ 0x08
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#define HARMONY_SR_16KHZ 0x09
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#define HARMONY_SR_27KHZ 0x0A
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#define HARMONY_SR_32KHZ 0x0B
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#define HARMONY_SR_48KHZ 0x0E
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#define HARMONY_SR_9KHZ 0x0F
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#define HARMONY_SR_5KHZ 0x10
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#define HARMONY_SR_11KHZ 0x11
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#define HARMONY_SR_18KHZ 0x12
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#define HARMONY_SR_22KHZ 0x13
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#define HARMONY_SR_37KHZ 0x14
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#define HARMONY_SR_44KHZ 0x15
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#define HARMONY_SR_33KHZ 0x16
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#define HARMONY_SR_6KHZ 0x17
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/*
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* Some magics numbers used to auto-detect file formats
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*/
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#define HARMONY_MAGIC_8B_ULAW 1
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#define HARMONY_MAGIC_8B_ALAW 27
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#define HARMONY_MAGIC_16B_LINEAR 3
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#define HARMONY_MAGIC_MONO 1
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#define HARMONY_MAGIC_STEREO 2
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/*
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* Channels Positions in mixer register
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*/
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#define GAIN_HE_SHIFT 27
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#define GAIN_HE_MASK ( 1 << GAIN_HE_SHIFT)
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#define GAIN_LE_SHIFT 26
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#define GAIN_LE_MASK ( 1 << GAIN_LE_SHIFT)
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#define GAIN_SE_SHIFT 25
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#define GAIN_SE_MASK ( 1 << GAIN_SE_SHIFT)
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#define GAIN_IS_SHIFT 24
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#define GAIN_IS_MASK ( 1 << GAIN_IS_SHIFT)
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#define GAIN_MA_SHIFT 20
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#define GAIN_MA_MASK ( 0x0f << GAIN_MA_SHIFT)
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#define GAIN_LI_SHIFT 16
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#define GAIN_LI_MASK ( 0x0f << GAIN_LI_SHIFT)
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#define GAIN_RI_SHIFT 12
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#define GAIN_RI_MASK ( 0x0f << GAIN_RI_SHIFT)
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#define GAIN_LO_SHIFT 6
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#define GAIN_LO_MASK ( 0x3f << GAIN_LO_SHIFT)
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#define GAIN_RO_SHIFT 0
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#define GAIN_RO_MASK ( 0x3f << GAIN_RO_SHIFT)
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#define MAX_OUTPUT_LEVEL (GAIN_RO_MASK >> GAIN_RO_SHIFT)
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#define MAX_INPUT_LEVEL (GAIN_RI_MASK >> GAIN_RI_SHIFT)
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#define MAX_MONITOR_LEVEL (GAIN_MA_MASK >> GAIN_MA_SHIFT)
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#define MIXER_INTERNAL SOUND_MIXER_LINE1
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#define MIXER_LINEOUT SOUND_MIXER_LINE2
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#define MIXER_HEADPHONES SOUND_MIXER_LINE3
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#define MASK_INTERNAL SOUND_MASK_LINE1
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#define MASK_LINEOUT SOUND_MASK_LINE2
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#define MASK_HEADPHONES SOUND_MASK_LINE3
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/*
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* Channels Mask in mixer register
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*/
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#define GAIN_TOTAL_SILENCE 0x00F00FFF
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#define GAIN_DEFAULT 0x0FF00000
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struct harmony_hpa {
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u8 unused000;
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u8 id;
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u8 teleshare_id;
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u8 unused003;
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u32 reset;
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u32 cntl;
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u32 gainctl;
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u32 pnxtadd;
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u32 pcuradd;
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u32 rnxtadd;
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u32 rcuradd;
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u32 dstatus;
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u32 ov;
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u32 pio;
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u32 unused02c;
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u32 unused030[3];
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u32 diag;
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};
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struct harmony_dev {
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struct harmony_hpa *hpa;
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struct parisc_device *dev;
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u32 current_gain;
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u32 dac_rate; /* 8000 ... 48000 (Hz) */
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u8 data_format; /* HARMONY_DF_xx_BIT_xxx */
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u8 sample_rate; /* HARMONY_SR_xx_KHZ */
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u8 stereo_select; /* HARMONY_SS_MONO or HARMONY_SS_STEREO */
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int format_initialized :1;
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int suspended_playing :1;
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int suspended_recording :1;
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int blocked_playing :1;
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int blocked_recording :1;
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int audio_open :1;
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int mixer_open :1;
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wait_queue_head_t wq_play, wq_record;
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int first_filled_play; /* first buffer containing data (next to play) */
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int nb_filled_play;
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int play_offset;
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int first_filled_record;
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int nb_filled_record;
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int dsp_unit, mixer_unit;
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};
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static struct harmony_dev harmony;
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/*
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* Dynamic sound buffer allocation and DMA memory
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*/
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struct harmony_buffer {
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unsigned char *addr;
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dma_addr_t dma_handle;
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int dma_coherent; /* Zero if dma_alloc_coherent() fails */
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unsigned int len;
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};
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/*
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* Harmony memory buffers
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*/
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static struct harmony_buffer played_buf, recorded_buf, silent, graveyard;
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#define CHECK_WBACK_INV_OFFSET(b,offset,len) \
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do { if (!b.dma_coherent) \
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dma_cache_wback_inv((unsigned long)b.addr+offset,len); \
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} while (0)
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static int __init harmony_alloc_buffer(struct harmony_buffer *b,
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unsigned int buffer_count)
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{
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b->len = buffer_count * HARMONY_BUF_SIZE;
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b->addr = dma_alloc_coherent(&harmony.dev->dev,
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b->len, &b->dma_handle, GFP_KERNEL|GFP_DMA);
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if (b->addr && b->dma_handle) {
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b->dma_coherent = 1;
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DPRINTK(KERN_INFO PFX "coherent memory: 0x%lx, played_buf: 0x%lx\n",
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(unsigned long)b->dma_handle, (unsigned long)b->addr);
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} else {
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b->dma_coherent = 0;
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/* kmalloc()ed memory will HPMC on ccio machines ! */
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b->addr = kmalloc(b->len, GFP_KERNEL);
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if (!b->addr) {
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printk(KERN_ERR PFX "couldn't allocate memory\n");
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return -EBUSY;
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}
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b->dma_handle = __pa(b->addr);
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}
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return 0;
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}
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static void __exit harmony_free_buffer(struct harmony_buffer *b)
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{
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if (!b->addr)
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return;
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if (b->dma_coherent)
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dma_free_coherent(&harmony.dev->dev,
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b->len, b->addr, b->dma_handle);
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else
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kfree(b->addr);
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memset(b, 0, sizeof(*b));
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}
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/*
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* Low-Level sound-chip programming
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*/
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static void __inline__ harmony_wait_CNTL(void)
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{
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/* Wait until we're out of control mode */
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while (gsc_readl(&harmony.hpa->cntl) & CNTL_C)
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/* wait */ ;
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}
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static void harmony_update_control(void)
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{
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u32 default_cntl;
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/* Set CNTL */
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default_cntl = (CNTL_C | /* The C bit */
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(harmony.data_format << 6) | /* Set the data format */
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(harmony.stereo_select << 5) | /* Stereo select */
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(harmony.sample_rate)); /* Set sample rate */
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harmony.format_initialized = 1;
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/* initialize CNTL */
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gsc_writel(default_cntl, &harmony.hpa->cntl);
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}
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static void harmony_set_control(u8 data_format, u8 sample_rate, u8 stereo_select)
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{
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harmony.sample_rate = sample_rate;
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harmony.data_format = data_format;
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harmony.stereo_select = stereo_select;
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harmony_update_control();
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}
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static void harmony_set_rate(u8 data_rate)
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{
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harmony.sample_rate = data_rate;
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harmony_update_control();
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}
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static int harmony_detect_rate(int *freq)
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{
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int newrate;
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switch (*freq) {
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case 8000: newrate = HARMONY_SR_8KHZ; break;
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case 16000: newrate = HARMONY_SR_16KHZ; break;
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case 27428: newrate = HARMONY_SR_27KHZ; break;
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case 32000: newrate = HARMONY_SR_32KHZ; break;
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case 48000: newrate = HARMONY_SR_48KHZ; break;
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case 9600: newrate = HARMONY_SR_9KHZ; break;
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case 5512: newrate = HARMONY_SR_5KHZ; break;
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case 11025: newrate = HARMONY_SR_11KHZ; break;
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case 18900: newrate = HARMONY_SR_18KHZ; break;
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case 22050: newrate = HARMONY_SR_22KHZ; break;
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case 37800: newrate = HARMONY_SR_37KHZ; break;
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case 44100: newrate = HARMONY_SR_44KHZ; break;
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case 33075: newrate = HARMONY_SR_33KHZ; break;
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case 6615: newrate = HARMONY_SR_6KHZ; break;
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default: newrate = HARMONY_SR_8KHZ;
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*freq = 8000; break;
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}
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return newrate;
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}
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static void harmony_set_format(u8 data_format)
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{
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harmony.data_format = data_format;
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harmony_update_control();
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}
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static void harmony_set_stereo(u8 stereo_select)
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{
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harmony.stereo_select = stereo_select;
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harmony_update_control();
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}
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static void harmony_disable_interrupts(void)
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{
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harmony_wait_CNTL();
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gsc_writel(0, &harmony.hpa->dstatus);
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}
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static void harmony_enable_interrupts(void)
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{
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harmony_wait_CNTL();
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gsc_writel(DSTATUS_IE, &harmony.hpa->dstatus);
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}
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/*
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* harmony_silence()
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*
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* This subroutine fills in a buffer starting at location start and
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* silences for length bytes. This references the current
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* configuration of the audio format.
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*
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*/
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static void harmony_silence(struct harmony_buffer *buffer, int start, int length)
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{
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u8 silence_char;
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/* Despite what you hear, silence is different in
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different audio formats. */
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switch (harmony.data_format) {
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case HARMONY_DF_8BIT_ULAW: silence_char = 0x55; break;
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case HARMONY_DF_8BIT_ALAW: silence_char = 0xff; break;
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case HARMONY_DF_16BIT_LINEAR: /* fall through */
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default: silence_char = 0;
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}
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memset(buffer->addr+start, silence_char, length);
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}
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static int harmony_audio_open(struct inode *inode, struct file *file)
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{
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if (harmony.audio_open)
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return -EBUSY;
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harmony.audio_open = 1;
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harmony.suspended_playing = harmony.suspended_recording = 1;
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harmony.blocked_playing = harmony.blocked_recording = 0;
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harmony.first_filled_play = harmony.first_filled_record = 0;
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harmony.nb_filled_play = harmony.nb_filled_record = 0;
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harmony.play_offset = 0;
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init_waitqueue_head(&harmony.wq_play);
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init_waitqueue_head(&harmony.wq_record);
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/* Start off in a balanced mode. */
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harmony_set_control(HARMONY_DF_8BIT_ULAW, HARMONY_SR_8KHZ, HARMONY_SS_MONO);
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harmony_update_control();
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harmony.format_initialized = 0;
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/* Clear out all the buffers and flush to cache */
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harmony_silence(&played_buf, 0, HARMONY_BUF_SIZE*MAX_BUFS);
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CHECK_WBACK_INV_OFFSET(played_buf, 0, HARMONY_BUF_SIZE*MAX_BUFS);
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return 0;
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}
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/*
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* Release (close) the audio device.
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*/
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static int harmony_audio_release(struct inode *inode, struct file *file)
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{
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if (!harmony.audio_open)
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return -EBUSY;
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harmony.audio_open = 0;
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return 0;
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}
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/*
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* Read recorded data off the audio device.
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*/
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static ssize_t harmony_audio_read(struct file *file,
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char *buffer,
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size_t size_count,
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loff_t *ppos)
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{
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int total_count = (int) size_count;
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int count = 0;
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int buf_to_read;
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while (count<total_count) {
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/* Wait until we're out of control mode */
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harmony_wait_CNTL();
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|
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/* Figure out which buffer to fill in */
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if (harmony.nb_filled_record <= 2) {
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harmony.blocked_recording = 1;
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if (harmony.suspended_recording) {
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harmony.suspended_recording = 0;
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harmony_enable_interrupts();
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}
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interruptible_sleep_on(&harmony.wq_record);
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harmony.blocked_recording = 0;
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}
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if (harmony.nb_filled_record < 2)
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return -EBUSY;
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buf_to_read = harmony.first_filled_record;
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/* Copy the page to an aligned buffer */
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if (copy_to_user(buffer+count, recorded_buf.addr +
|
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(HARMONY_BUF_SIZE*buf_to_read),
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HARMONY_BUF_SIZE)) {
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count = -EFAULT;
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break;
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}
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harmony.nb_filled_record--;
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harmony.first_filled_record++;
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harmony.first_filled_record %= MAX_BUFS;
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count += HARMONY_BUF_SIZE;
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}
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return count;
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}
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/*
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* Here is the place where we try to recognize file format.
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* Sun/NeXT .au files begin with the string .snd
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* At offset 12 is specified the encoding.
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* At offset 16 is specified speed rate
|
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* At Offset 20 is specified the numbers of voices
|
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*/
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#define four_bytes_to_u32(start) (file_header[start] << 24)|\
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(file_header[start+1] << 16)|\
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(file_header[start+2] << 8)|\
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(file_header[start+3]);
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#define test_rate(tested,real_value,harmony_value) if ((tested)<=(real_value))\
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static int harmony_format_auto_detect(const char *buffer, int block_size)
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{
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u8 file_header[24];
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u32 start_string;
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int ret = 0;
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|
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if (block_size>24) {
|
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if (copy_from_user(file_header, buffer, sizeof(file_header)))
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ret = -EFAULT;
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start_string = four_bytes_to_u32(0);
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|
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if ((file_header[4]==0) && (start_string==0x2E736E64)) {
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u32 format;
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u32 nb_voices;
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u32 speed;
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format = four_bytes_to_u32(12);
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nb_voices = four_bytes_to_u32(20);
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speed = four_bytes_to_u32(16);
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switch (format) {
|
|
case HARMONY_MAGIC_8B_ULAW:
|
|
harmony.data_format = HARMONY_DF_8BIT_ULAW;
|
|
break;
|
|
case HARMONY_MAGIC_8B_ALAW:
|
|
harmony.data_format = HARMONY_DF_8BIT_ALAW;
|
|
break;
|
|
case HARMONY_MAGIC_16B_LINEAR:
|
|
harmony.data_format = HARMONY_DF_16BIT_LINEAR;
|
|
break;
|
|
default:
|
|
harmony_set_control(HARMONY_DF_16BIT_LINEAR,
|
|
HARMONY_SR_44KHZ, HARMONY_SS_STEREO);
|
|
goto out;
|
|
}
|
|
switch (nb_voices) {
|
|
case HARMONY_MAGIC_MONO:
|
|
harmony.stereo_select = HARMONY_SS_MONO;
|
|
break;
|
|
case HARMONY_MAGIC_STEREO:
|
|
harmony.stereo_select = HARMONY_SS_STEREO;
|
|
break;
|
|
default:
|
|
harmony.stereo_select = HARMONY_SS_MONO;
|
|
break;
|
|
}
|
|
harmony_set_rate(harmony_detect_rate(&speed));
|
|
harmony.dac_rate = speed;
|
|
goto out;
|
|
}
|
|
}
|
|
harmony_set_control(HARMONY_DF_8BIT_ULAW, HARMONY_SR_8KHZ, HARMONY_SS_MONO);
|
|
out:
|
|
return ret;
|
|
}
|
|
#undef four_bytes_to_u32
|
|
|
|
|
|
static ssize_t harmony_audio_write(struct file *file,
|
|
const char *buffer,
|
|
size_t size_count,
|
|
loff_t *ppos)
|
|
{
|
|
int total_count = (int) size_count;
|
|
int count = 0;
|
|
int frame_size;
|
|
int buf_to_fill;
|
|
int fresh_buffer;
|
|
|
|
if (!harmony.format_initialized) {
|
|
if (harmony_format_auto_detect(buffer, total_count))
|
|
return -EFAULT;
|
|
}
|
|
|
|
while (count<total_count) {
|
|
/* Wait until we're out of control mode */
|
|
harmony_wait_CNTL();
|
|
|
|
/* Figure out which buffer to fill in */
|
|
if (harmony.nb_filled_play+2 >= MAX_BUFS && !harmony.play_offset) {
|
|
harmony.blocked_playing = 1;
|
|
interruptible_sleep_on(&harmony.wq_play);
|
|
harmony.blocked_playing = 0;
|
|
}
|
|
if (harmony.nb_filled_play+2 >= MAX_BUFS && !harmony.play_offset)
|
|
return -EBUSY;
|
|
|
|
|
|
buf_to_fill = (harmony.first_filled_play+harmony.nb_filled_play);
|
|
if (harmony.play_offset) {
|
|
buf_to_fill--;
|
|
buf_to_fill += MAX_BUFS;
|
|
}
|
|
buf_to_fill %= MAX_BUFS;
|
|
|
|
fresh_buffer = (harmony.play_offset == 0);
|
|
|
|
/* Figure out the size of the frame */
|
|
if ((total_count-count) >= HARMONY_BUF_SIZE - harmony.play_offset) {
|
|
frame_size = HARMONY_BUF_SIZE - harmony.play_offset;
|
|
} else {
|
|
frame_size = total_count - count;
|
|
/* Clear out the buffer, since there we'll only be
|
|
overlaying part of the old buffer with the new one */
|
|
harmony_silence(&played_buf,
|
|
HARMONY_BUF_SIZE*buf_to_fill+frame_size+harmony.play_offset,
|
|
HARMONY_BUF_SIZE-frame_size-harmony.play_offset);
|
|
}
|
|
|
|
/* Copy the page to an aligned buffer */
|
|
if (copy_from_user(played_buf.addr +(HARMONY_BUF_SIZE*buf_to_fill) + harmony.play_offset,
|
|
buffer+count, frame_size))
|
|
return -EFAULT;
|
|
CHECK_WBACK_INV_OFFSET(played_buf, (HARMONY_BUF_SIZE*buf_to_fill + harmony.play_offset),
|
|
frame_size);
|
|
|
|
if (fresh_buffer)
|
|
harmony.nb_filled_play++;
|
|
|
|
count += frame_size;
|
|
harmony.play_offset += frame_size;
|
|
harmony.play_offset %= HARMONY_BUF_SIZE;
|
|
if (harmony.suspended_playing && (harmony.nb_filled_play>=4))
|
|
harmony_enable_interrupts();
|
|
}
|
|
|
|
return count;
|
|
}
|
|
|
|
static unsigned int harmony_audio_poll(struct file *file,
|
|
struct poll_table_struct *wait)
|
|
{
|
|
unsigned int mask = 0;
|
|
|
|
if (file->f_mode & FMODE_READ) {
|
|
if (!harmony.suspended_recording)
|
|
poll_wait(file, &harmony.wq_record, wait);
|
|
if (harmony.nb_filled_record)
|
|
mask |= POLLIN | POLLRDNORM;
|
|
}
|
|
|
|
if (file->f_mode & FMODE_WRITE) {
|
|
if (!harmony.suspended_playing)
|
|
poll_wait(file, &harmony.wq_play, wait);
|
|
if (harmony.nb_filled_play)
|
|
mask |= POLLOUT | POLLWRNORM;
|
|
}
|
|
|
|
return mask;
|
|
}
|
|
|
|
static int harmony_audio_ioctl(struct inode *inode,
|
|
struct file *file,
|
|
unsigned int cmd,
|
|
unsigned long arg)
|
|
{
|
|
int ival, new_format;
|
|
int frag_size, frag_buf;
|
|
struct audio_buf_info info;
|
|
|
|
switch (cmd) {
|
|
case OSS_GETVERSION:
|
|
return put_user(SOUND_VERSION, (int *) arg);
|
|
|
|
case SNDCTL_DSP_GETCAPS:
|
|
ival = DSP_CAP_DUPLEX;
|
|
return put_user(ival, (int *) arg);
|
|
|
|
case SNDCTL_DSP_GETFMTS:
|
|
ival = (AFMT_S16_BE | AFMT_MU_LAW | AFMT_A_LAW );
|
|
return put_user(ival, (int *) arg);
|
|
|
|
case SNDCTL_DSP_SETFMT:
|
|
if (get_user(ival, (int *) arg))
|
|
return -EFAULT;
|
|
if (ival != AFMT_QUERY) {
|
|
switch (ival) {
|
|
case AFMT_MU_LAW: new_format = HARMONY_DF_8BIT_ULAW; break;
|
|
case AFMT_A_LAW: new_format = HARMONY_DF_8BIT_ALAW; break;
|
|
case AFMT_S16_BE: new_format = HARMONY_DF_16BIT_LINEAR; break;
|
|
default: {
|
|
DPRINTK(KERN_WARNING PFX
|
|
"unsupported sound format 0x%04x requested.\n",
|
|
ival);
|
|
ival = AFMT_S16_BE;
|
|
return put_user(ival, (int *) arg);
|
|
}
|
|
}
|
|
harmony_set_format(new_format);
|
|
return 0;
|
|
} else {
|
|
switch (harmony.data_format) {
|
|
case HARMONY_DF_8BIT_ULAW: ival = AFMT_MU_LAW; break;
|
|
case HARMONY_DF_8BIT_ALAW: ival = AFMT_A_LAW; break;
|
|
case HARMONY_DF_16BIT_LINEAR: ival = AFMT_U16_BE; break;
|
|
default: ival = 0;
|
|
}
|
|
return put_user(ival, (int *) arg);
|
|
}
|
|
|
|
case SOUND_PCM_READ_RATE:
|
|
ival = harmony.dac_rate;
|
|
return put_user(ival, (int *) arg);
|
|
|
|
case SNDCTL_DSP_SPEED:
|
|
if (get_user(ival, (int *) arg))
|
|
return -EFAULT;
|
|
harmony_set_rate(harmony_detect_rate(&ival));
|
|
harmony.dac_rate = ival;
|
|
return put_user(ival, (int*) arg);
|
|
|
|
case SNDCTL_DSP_STEREO:
|
|
if (get_user(ival, (int *) arg))
|
|
return -EFAULT;
|
|
if (ival != 0 && ival != 1)
|
|
return -EINVAL;
|
|
harmony_set_stereo(ival);
|
|
return 0;
|
|
|
|
case SNDCTL_DSP_CHANNELS:
|
|
if (get_user(ival, (int *) arg))
|
|
return -EFAULT;
|
|
if (ival != 1 && ival != 2) {
|
|
ival = harmony.stereo_select == HARMONY_SS_MONO ? 1 : 2;
|
|
return put_user(ival, (int *) arg);
|
|
}
|
|
harmony_set_stereo(ival-1);
|
|
return 0;
|
|
|
|
case SNDCTL_DSP_GETBLKSIZE:
|
|
ival = HARMONY_BUF_SIZE;
|
|
return put_user(ival, (int *) arg);
|
|
|
|
case SNDCTL_DSP_NONBLOCK:
|
|
file->f_flags |= O_NONBLOCK;
|
|
return 0;
|
|
|
|
case SNDCTL_DSP_RESET:
|
|
if (!harmony.suspended_recording) {
|
|
/* TODO: stop_recording() */
|
|
}
|
|
return 0;
|
|
|
|
case SNDCTL_DSP_SETFRAGMENT:
|
|
if (get_user(ival, (int *)arg))
|
|
return -EFAULT;
|
|
frag_size = ival & 0xffff;
|
|
frag_buf = (ival>>16) & 0xffff;
|
|
/* TODO: We use hardcoded fragment sizes and numbers for now */
|
|
frag_size = 12; /* 4096 == 2^12 */
|
|
frag_buf = MAX_BUFS;
|
|
ival = (frag_buf << 16) + frag_size;
|
|
return put_user(ival, (int *) arg);
|
|
|
|
case SNDCTL_DSP_GETOSPACE:
|
|
if (!(file->f_mode & FMODE_WRITE))
|
|
return -EINVAL;
|
|
info.fragstotal = MAX_BUFS;
|
|
info.fragments = MAX_BUFS - harmony.nb_filled_play;
|
|
info.fragsize = HARMONY_BUF_SIZE;
|
|
info.bytes = info.fragments * info.fragsize;
|
|
return copy_to_user((void *)arg, &info, sizeof(info)) ? -EFAULT : 0;
|
|
|
|
case SNDCTL_DSP_GETISPACE:
|
|
if (!(file->f_mode & FMODE_READ))
|
|
return -EINVAL;
|
|
info.fragstotal = MAX_BUFS;
|
|
info.fragments = /*MAX_BUFS-*/ harmony.nb_filled_record;
|
|
info.fragsize = HARMONY_BUF_SIZE;
|
|
info.bytes = info.fragments * info.fragsize;
|
|
return copy_to_user((void *)arg, &info, sizeof(info)) ? -EFAULT : 0;
|
|
|
|
case SNDCTL_DSP_SYNC:
|
|
return 0;
|
|
}
|
|
|
|
return -EINVAL;
|
|
}
|
|
|
|
|
|
/*
|
|
* harmony_interrupt()
|
|
*
|
|
* harmony interruption service routine
|
|
*
|
|
*/
|
|
|
|
static irqreturn_t harmony_interrupt(int irq, void *dev, struct pt_regs *regs)
|
|
{
|
|
u32 dstatus;
|
|
struct harmony_hpa *hpa;
|
|
|
|
/* Setup the hpa */
|
|
hpa = ((struct harmony_dev *)dev)->hpa;
|
|
harmony_wait_CNTL();
|
|
|
|
/* Read dstatus and pcuradd (the current address) */
|
|
dstatus = gsc_readl(&hpa->dstatus);
|
|
|
|
/* Turn off interrupts */
|
|
harmony_disable_interrupts();
|
|
|
|
/* Check if this is a request to get the next play buffer */
|
|
if (dstatus & DSTATUS_PN) {
|
|
if (!harmony.nb_filled_play) {
|
|
harmony.suspended_playing = 1;
|
|
gsc_writel((unsigned long)silent.dma_handle, &hpa->pnxtadd);
|
|
|
|
if (!harmony.suspended_recording)
|
|
harmony_enable_interrupts();
|
|
} else {
|
|
harmony.suspended_playing = 0;
|
|
gsc_writel((unsigned long)played_buf.dma_handle +
|
|
(HARMONY_BUF_SIZE*harmony.first_filled_play),
|
|
&hpa->pnxtadd);
|
|
harmony.first_filled_play++;
|
|
harmony.first_filled_play %= MAX_BUFS;
|
|
harmony.nb_filled_play--;
|
|
|
|
harmony_enable_interrupts();
|
|
}
|
|
|
|
if (harmony.blocked_playing)
|
|
wake_up_interruptible(&harmony.wq_play);
|
|
}
|
|
|
|
/* Check if we're being asked to fill in a recording buffer */
|
|
if (dstatus & DSTATUS_RN) {
|
|
if((harmony.nb_filled_record+2>=MAX_BUFS) || harmony.suspended_recording)
|
|
{
|
|
harmony.nb_filled_record = 0;
|
|
harmony.first_filled_record = 0;
|
|
harmony.suspended_recording = 1;
|
|
gsc_writel((unsigned long)graveyard.dma_handle, &hpa->rnxtadd);
|
|
if (!harmony.suspended_playing)
|
|
harmony_enable_interrupts();
|
|
} else {
|
|
int buf_to_fill;
|
|
buf_to_fill = (harmony.first_filled_record+harmony.nb_filled_record) % MAX_BUFS;
|
|
CHECK_WBACK_INV_OFFSET(recorded_buf, HARMONY_BUF_SIZE*buf_to_fill, HARMONY_BUF_SIZE);
|
|
gsc_writel((unsigned long)recorded_buf.dma_handle +
|
|
HARMONY_BUF_SIZE*buf_to_fill,
|
|
&hpa->rnxtadd);
|
|
harmony.nb_filled_record++;
|
|
harmony_enable_interrupts();
|
|
}
|
|
|
|
if (harmony.blocked_recording && harmony.nb_filled_record>3)
|
|
wake_up_interruptible(&harmony.wq_record);
|
|
}
|
|
return IRQ_HANDLED;
|
|
}
|
|
|
|
/*
|
|
* Sound playing functions
|
|
*/
|
|
|
|
static struct file_operations harmony_audio_fops = {
|
|
.owner = THIS_MODULE,
|
|
.llseek = no_llseek,
|
|
.read = harmony_audio_read,
|
|
.write = harmony_audio_write,
|
|
.poll = harmony_audio_poll,
|
|
.ioctl = harmony_audio_ioctl,
|
|
.open = harmony_audio_open,
|
|
.release = harmony_audio_release,
|
|
};
|
|
|
|
static int harmony_audio_init(void)
|
|
{
|
|
/* Request that IRQ */
|
|
if (request_irq(harmony.dev->irq, harmony_interrupt, 0 ,"harmony", &harmony)) {
|
|
printk(KERN_ERR PFX "Error requesting irq %d.\n", harmony.dev->irq);
|
|
return -EFAULT;
|
|
}
|
|
|
|
harmony.dsp_unit = register_sound_dsp(&harmony_audio_fops, -1);
|
|
if (harmony.dsp_unit < 0) {
|
|
printk(KERN_ERR PFX "Error registering dsp\n");
|
|
free_irq(harmony.dev->irq, &harmony);
|
|
return -EFAULT;
|
|
}
|
|
|
|
/* Clear the buffers so you don't end up with crap in the buffers. */
|
|
harmony_silence(&played_buf, 0, HARMONY_BUF_SIZE*MAX_BUFS);
|
|
|
|
/* Make sure this makes it to cache */
|
|
CHECK_WBACK_INV_OFFSET(played_buf, 0, HARMONY_BUF_SIZE*MAX_BUFS);
|
|
|
|
/* Clear out the silent buffer and flush to cache */
|
|
harmony_silence(&silent, 0, HARMONY_BUF_SIZE);
|
|
CHECK_WBACK_INV_OFFSET(silent, 0, HARMONY_BUF_SIZE);
|
|
|
|
harmony.audio_open = 0;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/*
|
|
* mixer functions
|
|
*/
|
|
|
|
static void harmony_mixer_set_gain(void)
|
|
{
|
|
harmony_wait_CNTL();
|
|
gsc_writel(harmony.current_gain, &harmony.hpa->gainctl);
|
|
}
|
|
|
|
/*
|
|
* Read gain of selected channel.
|
|
* The OSS rate is from 0 (silent) to 100 -> need some conversions
|
|
*
|
|
* The harmony gain are attenuation for output and monitor gain.
|
|
* is amplifaction for input gain
|
|
*/
|
|
#define to_harmony_level(level,max) ((level)*max/100)
|
|
#define to_oss_level(level,max) ((level)*100/max)
|
|
|
|
static int harmony_mixer_get_level(int channel)
|
|
{
|
|
int left_level;
|
|
int right_level;
|
|
|
|
switch (channel) {
|
|
case SOUND_MIXER_VOLUME:
|
|
left_level = (harmony.current_gain & GAIN_LO_MASK) >> GAIN_LO_SHIFT;
|
|
right_level = (harmony.current_gain & GAIN_RO_MASK) >> GAIN_RO_SHIFT;
|
|
left_level = to_oss_level(MAX_OUTPUT_LEVEL - left_level, MAX_OUTPUT_LEVEL);
|
|
right_level = to_oss_level(MAX_OUTPUT_LEVEL - right_level, MAX_OUTPUT_LEVEL);
|
|
return (right_level << 8)+left_level;
|
|
|
|
case SOUND_MIXER_IGAIN:
|
|
left_level = (harmony.current_gain & GAIN_LI_MASK) >> GAIN_LI_SHIFT;
|
|
right_level= (harmony.current_gain & GAIN_RI_MASK) >> GAIN_RI_SHIFT;
|
|
left_level = to_oss_level(left_level, MAX_INPUT_LEVEL);
|
|
right_level= to_oss_level(right_level, MAX_INPUT_LEVEL);
|
|
return (right_level << 8)+left_level;
|
|
|
|
case SOUND_MIXER_MONITOR:
|
|
left_level = (harmony.current_gain & GAIN_MA_MASK) >> GAIN_MA_SHIFT;
|
|
left_level = to_oss_level(MAX_MONITOR_LEVEL-left_level, MAX_MONITOR_LEVEL);
|
|
return (left_level << 8)+left_level;
|
|
}
|
|
return -EINVAL;
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
* Some conversions for the same reasons.
|
|
* We give back the new real value(s) due to
|
|
* the rescale.
|
|
*/
|
|
|
|
static int harmony_mixer_set_level(int channel, int value)
|
|
{
|
|
int left_level;
|
|
int right_level;
|
|
int new_left_level;
|
|
int new_right_level;
|
|
|
|
right_level = (value & 0x0000ff00) >> 8;
|
|
left_level = value & 0x000000ff;
|
|
if (right_level > 100) right_level = 100;
|
|
if (left_level > 100) left_level = 100;
|
|
|
|
switch (channel) {
|
|
case SOUND_MIXER_VOLUME:
|
|
right_level = to_harmony_level(100-right_level, MAX_OUTPUT_LEVEL);
|
|
left_level = to_harmony_level(100-left_level, MAX_OUTPUT_LEVEL);
|
|
new_right_level = to_oss_level(MAX_OUTPUT_LEVEL - right_level, MAX_OUTPUT_LEVEL);
|
|
new_left_level = to_oss_level(MAX_OUTPUT_LEVEL - left_level, MAX_OUTPUT_LEVEL);
|
|
harmony.current_gain = (harmony.current_gain & ~(GAIN_LO_MASK | GAIN_RO_MASK))
|
|
| (left_level << GAIN_LO_SHIFT) | (right_level << GAIN_RO_SHIFT);
|
|
harmony_mixer_set_gain();
|
|
return (new_right_level << 8) + new_left_level;
|
|
|
|
case SOUND_MIXER_IGAIN:
|
|
right_level = to_harmony_level(right_level, MAX_INPUT_LEVEL);
|
|
left_level = to_harmony_level(left_level, MAX_INPUT_LEVEL);
|
|
new_right_level = to_oss_level(right_level, MAX_INPUT_LEVEL);
|
|
new_left_level = to_oss_level(left_level, MAX_INPUT_LEVEL);
|
|
harmony.current_gain = (harmony.current_gain & ~(GAIN_LI_MASK | GAIN_RI_MASK))
|
|
| (left_level << GAIN_LI_SHIFT) | (right_level << GAIN_RI_SHIFT);
|
|
harmony_mixer_set_gain();
|
|
return (new_right_level << 8) + new_left_level;
|
|
|
|
case SOUND_MIXER_MONITOR:
|
|
left_level = to_harmony_level(100-left_level, MAX_MONITOR_LEVEL);
|
|
new_left_level = to_oss_level(MAX_MONITOR_LEVEL-left_level, MAX_MONITOR_LEVEL);
|
|
harmony.current_gain = (harmony.current_gain & ~GAIN_MA_MASK) | (left_level << GAIN_MA_SHIFT);
|
|
harmony_mixer_set_gain();
|
|
return (new_left_level << 8) + new_left_level;
|
|
}
|
|
|
|
return -EINVAL;
|
|
}
|
|
|
|
#undef to_harmony_level
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#undef to_oss_level
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/*
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* Return the selected input device (mic or line)
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*/
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static int harmony_mixer_get_recmask(void)
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{
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int current_input_line;
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current_input_line = (harmony.current_gain & GAIN_IS_MASK)
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>> GAIN_IS_SHIFT;
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if (current_input_line)
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return SOUND_MASK_MIC;
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return SOUND_MASK_LINE;
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}
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/*
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* Set the input (only one at time, arbitrary priority to line in)
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*/
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static int harmony_mixer_set_recmask(int recmask)
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{
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int new_input_line;
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int new_input_mask;
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int current_input_line;
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current_input_line = (harmony.current_gain & GAIN_IS_MASK)
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>> GAIN_IS_SHIFT;
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if ((current_input_line && ((recmask & SOUND_MASK_LINE) || !(recmask & SOUND_MASK_MIC))) ||
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(!current_input_line && ((recmask & SOUND_MASK_LINE) && !(recmask & SOUND_MASK_MIC)))) {
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new_input_line = 0;
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new_input_mask = SOUND_MASK_LINE;
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} else {
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new_input_line = 1;
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new_input_mask = SOUND_MASK_MIC;
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}
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harmony.current_gain = ((harmony.current_gain & ~GAIN_IS_MASK) |
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(new_input_line << GAIN_IS_SHIFT ));
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harmony_mixer_set_gain();
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return new_input_mask;
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}
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/*
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* give the active outlines
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*/
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static int harmony_mixer_get_outmask(void)
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{
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int outmask = 0;
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if (harmony.current_gain & GAIN_SE_MASK) outmask |= MASK_INTERNAL;
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if (harmony.current_gain & GAIN_LE_MASK) outmask |= MASK_LINEOUT;
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if (harmony.current_gain & GAIN_HE_MASK) outmask |= MASK_HEADPHONES;
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return outmask;
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}
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static int harmony_mixer_set_outmask(int outmask)
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{
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if (outmask & MASK_INTERNAL)
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harmony.current_gain |= GAIN_SE_MASK;
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else
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harmony.current_gain &= ~GAIN_SE_MASK;
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if (outmask & MASK_LINEOUT)
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harmony.current_gain |= GAIN_LE_MASK;
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else
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harmony.current_gain &= ~GAIN_LE_MASK;
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if (outmask & MASK_HEADPHONES)
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harmony.current_gain |= GAIN_HE_MASK;
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else
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harmony.current_gain &= ~GAIN_HE_MASK;
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harmony_mixer_set_gain();
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return (outmask & (MASK_INTERNAL | MASK_LINEOUT | MASK_HEADPHONES));
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}
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/*
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* This code is inspired from sb_mixer.c
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*/
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static int harmony_mixer_ioctl(struct inode * inode, struct file * file,
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unsigned int cmd, unsigned long arg)
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{
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int val;
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int ret;
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if (cmd == SOUND_MIXER_INFO) {
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mixer_info info;
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memset(&info, 0, sizeof(info));
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strncpy(info.id, "harmony", sizeof(info.id)-1);
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strncpy(info.name, "Harmony audio", sizeof(info.name)-1);
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info.modify_counter = 1; /* ? */
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if (copy_to_user((void *)arg, &info, sizeof(info)))
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return -EFAULT;
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return 0;
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}
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if (cmd == OSS_GETVERSION)
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return put_user(SOUND_VERSION, (int *)arg);
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/* read */
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val = 0;
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if (_SIOC_DIR(cmd) & _SIOC_WRITE)
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if (get_user(val, (int *)arg))
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return -EFAULT;
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switch (cmd) {
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case MIXER_READ(SOUND_MIXER_CAPS):
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ret = SOUND_CAP_EXCL_INPUT;
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break;
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case MIXER_READ(SOUND_MIXER_STEREODEVS):
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ret = SOUND_MASK_VOLUME | SOUND_MASK_IGAIN;
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break;
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case MIXER_READ(SOUND_MIXER_RECMASK):
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ret = SOUND_MASK_MIC | SOUND_MASK_LINE;
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break;
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case MIXER_READ(SOUND_MIXER_DEVMASK):
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ret = SOUND_MASK_VOLUME | SOUND_MASK_IGAIN |
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SOUND_MASK_MONITOR;
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break;
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case MIXER_READ(SOUND_MIXER_OUTMASK):
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ret = MASK_INTERNAL | MASK_LINEOUT |
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MASK_HEADPHONES;
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break;
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case MIXER_WRITE(SOUND_MIXER_RECSRC):
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ret = harmony_mixer_set_recmask(val);
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break;
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case MIXER_READ(SOUND_MIXER_RECSRC):
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ret = harmony_mixer_get_recmask();
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break;
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case MIXER_WRITE(SOUND_MIXER_OUTSRC):
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ret = harmony_mixer_set_outmask(val);
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break;
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case MIXER_READ(SOUND_MIXER_OUTSRC):
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ret = harmony_mixer_get_outmask();
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break;
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case MIXER_WRITE(SOUND_MIXER_VOLUME):
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case MIXER_WRITE(SOUND_MIXER_IGAIN):
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case MIXER_WRITE(SOUND_MIXER_MONITOR):
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ret = harmony_mixer_set_level(cmd & 0xff, val);
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break;
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case MIXER_READ(SOUND_MIXER_VOLUME):
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case MIXER_READ(SOUND_MIXER_IGAIN):
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case MIXER_READ(SOUND_MIXER_MONITOR):
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ret = harmony_mixer_get_level(cmd & 0xff);
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break;
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default:
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return -EINVAL;
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}
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if (put_user(ret, (int *)arg))
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return -EFAULT;
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return 0;
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}
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static int harmony_mixer_open(struct inode *inode, struct file *file)
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{
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if (harmony.mixer_open)
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return -EBUSY;
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harmony.mixer_open = 1;
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return 0;
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}
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static int harmony_mixer_release(struct inode *inode, struct file *file)
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{
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if (!harmony.mixer_open)
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return -EBUSY;
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harmony.mixer_open = 0;
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return 0;
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}
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static struct file_operations harmony_mixer_fops = {
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.owner = THIS_MODULE,
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.llseek = no_llseek,
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.open = harmony_mixer_open,
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.release = harmony_mixer_release,
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.ioctl = harmony_mixer_ioctl,
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};
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/*
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* Mute all the output and reset Harmony.
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*/
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static void __init harmony_mixer_reset(void)
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{
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harmony.current_gain = GAIN_TOTAL_SILENCE;
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harmony_mixer_set_gain();
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harmony_wait_CNTL();
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gsc_writel(1, &harmony.hpa->reset);
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mdelay(50); /* wait 50 ms */
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gsc_writel(0, &harmony.hpa->reset);
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harmony.current_gain = GAIN_DEFAULT;
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harmony_mixer_set_gain();
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}
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static int __init harmony_mixer_init(void)
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{
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/* Register the device file operations */
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harmony.mixer_unit = register_sound_mixer(&harmony_mixer_fops, -1);
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if (harmony.mixer_unit < 0) {
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printk(KERN_WARNING PFX "Error Registering Mixer Driver\n");
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return -EFAULT;
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}
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harmony_mixer_reset();
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harmony.mixer_open = 0;
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return 0;
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}
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/*
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* This is the callback that's called by the inventory hardware code
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* if it finds a match to the registered driver.
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*/
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static int __devinit
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harmony_driver_probe(struct parisc_device *dev)
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{
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u8 id;
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u8 rev;
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u32 cntl;
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int ret;
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if (harmony.hpa) {
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/* We only support one Harmony at this time */
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printk(KERN_ERR PFX "driver already registered\n");
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return -EBUSY;
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}
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if (!dev->irq) {
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printk(KERN_ERR PFX "no irq found\n");
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return -ENODEV;
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}
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/* Set the HPA of harmony */
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harmony.hpa = (struct harmony_hpa *)dev->hpa;
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harmony.dev = dev;
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/* Grab the ID and revision from the device */
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id = gsc_readb(&harmony.hpa->id);
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if ((id | 1) != 0x15) {
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printk(KERN_WARNING PFX "wrong harmony id 0x%02x\n", id);
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return -EBUSY;
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}
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cntl = gsc_readl(&harmony.hpa->cntl);
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rev = (cntl>>20) & 0xff;
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printk(KERN_INFO "Lasi Harmony Audio driver " HARMONY_VERSION ", "
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"h/w id %i, rev. %i at 0x%lx, IRQ %i\n",
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id, rev, dev->hpa, harmony.dev->irq);
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/* Make sure the control bit isn't set, although I don't think it
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ever is. */
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if (cntl & CNTL_C) {
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printk(KERN_WARNING PFX "CNTL busy\n");
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harmony.hpa = 0;
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return -EBUSY;
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}
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/* Initialize the memory buffers */
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if (harmony_alloc_buffer(&played_buf, MAX_BUFS) ||
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harmony_alloc_buffer(&recorded_buf, MAX_BUFS) ||
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harmony_alloc_buffer(&graveyard, 1) ||
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harmony_alloc_buffer(&silent, 1)) {
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ret = -EBUSY;
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goto out_err;
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}
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/* Initialize /dev/mixer and /dev/audio */
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if ((ret=harmony_mixer_init()))
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goto out_err;
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if ((ret=harmony_audio_init()))
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goto out_err;
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return 0;
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out_err:
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harmony.hpa = 0;
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harmony_free_buffer(&played_buf);
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harmony_free_buffer(&recorded_buf);
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harmony_free_buffer(&graveyard);
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harmony_free_buffer(&silent);
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return ret;
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}
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static struct parisc_device_id harmony_tbl[] = {
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/* { HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007A }, Bushmaster/Flounder */
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{ HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007B }, /* 712/715 Audio */
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{ HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007E }, /* Pace Audio */
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{ HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007F }, /* Outfield / Coral II */
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{ 0, }
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};
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MODULE_DEVICE_TABLE(parisc, harmony_tbl);
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static struct parisc_driver harmony_driver = {
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.name = "Lasi Harmony",
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.id_table = harmony_tbl,
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.probe = harmony_driver_probe,
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};
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static int __init init_harmony(void)
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{
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return register_parisc_driver(&harmony_driver);
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}
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static void __exit cleanup_harmony(void)
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{
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free_irq(harmony.dev->irq, &harmony);
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unregister_sound_mixer(harmony.mixer_unit);
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unregister_sound_dsp(harmony.dsp_unit);
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harmony_free_buffer(&played_buf);
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harmony_free_buffer(&recorded_buf);
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harmony_free_buffer(&graveyard);
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harmony_free_buffer(&silent);
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unregister_parisc_driver(&harmony_driver);
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}
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MODULE_AUTHOR("Alex DeVries <alex@onefishtwo.ca>");
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MODULE_DESCRIPTION("Harmony sound driver");
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MODULE_LICENSE("GPL");
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module_init(init_harmony);
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module_exit(cleanup_harmony);
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