/* *   SAA713x ALSA support for V4L * * *   Caveats: *        - Volume doesn't work (it's always at max) * *   This program is free software; you can redistribute it and/or modify *   it under the terms of the GNU General Public License as published by *   the Free Software Foundation, version 2 * *   This program is distributed in the hope that it will be useful, *   but WITHOUT ANY WARRANTY; without even the implied warranty of *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the *   GNU General Public License for more details. * *   You should have received a copy of the GNU General Public License *   along with this program; if not, write to the Free Software *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA * */#include <linux/init.h>#include <linux/slab.h>#include <linux/time.h>#include <linux/wait.h>#include <linux/moduleparam.h>#include <linux/module.h>#include <sound/driver.h>#include <sound/core.h>#include <sound/control.h>#include <sound/pcm.h>#include <sound/pcm_params.h>#include <sound/initval.h>#include <linux/interrupt.h>#include "saa7134.h"#include "saa7134-reg.h"static unsigned int debug  = 0;module_param(debug, int, 0644);MODULE_PARM_DESC(debug,"enable debug messages [alsa]");/* * Configuration macros *//* defaults */#define MIXER_ADDR_TVTUNER	0#define MIXER_ADDR_LINE1	1#define MIXER_ADDR_LINE2	2#define MIXER_ADDR_LAST		2static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 1};module_param_array(index, int, NULL, 0444);module_param_array(enable, int, NULL, 0444);MODULE_PARM_DESC(index, "Index value for SAA7134 capture interface(s).");MODULE_PARM_DESC(enable, "Enable (or not) the SAA7134 capture interface(s).");#define dprintk(fmt, arg...)    if (debug) \	printk(KERN_DEBUG "%s/alsa: " fmt, dev->name , ##arg)/* * Main chip structure */typedef struct snd_card_saa7134 {	snd_card_t *card;	spinlock_t mixer_lock;	int mixer_volume[MIXER_ADDR_LAST+1][2];	int capture_source[MIXER_ADDR_LAST+1][2];	struct pci_dev *pci;	struct saa7134_dev *dev;	unsigned long iobase;	int irq;	spinlock_t lock;} snd_card_saa7134_t;/* * PCM structure */typedef struct snd_card_saa7134_pcm {	struct saa7134_dev *dev;	spinlock_t lock;	snd_pcm_substream_t *substream;} snd_card_saa7134_pcm_t;static snd_card_t *snd_saa7134_cards[SNDRV_CARDS];/* * saa7134 DMA audio stop * *   Called when the capture device is released or the buffer overflows * *   - Copied verbatim from saa7134-oss's dsp_dma_stop. * */static void saa7134_dma_stop(struct saa7134_dev *dev){	dev->dmasound.dma_blk     = -1;	dev->dmasound.dma_running = 0;	saa7134_set_dmabits(dev);}/* * saa7134 DMA audio start * *   Called when preparing the capture device for use * *   - Copied verbatim from saa7134-oss's dsp_dma_start. * */static void saa7134_dma_start(struct saa7134_dev *dev){	dev->dmasound.dma_blk     = 0;	dev->dmasound.dma_running = 1;	saa7134_set_dmabits(dev);}/* * saa7134 audio DMA IRQ handler * *   Called whenever we get an SAA7134_IRQ_REPORT_DONE_RA3 interrupt *   Handles shifting between the 2 buffers, manages the read counters, *  and notifies ALSA when periods elapse * *   - Mostly copied from saa7134-oss's saa7134_irq_oss_done. * */static void saa7134_irq_alsa_done(struct saa7134_dev *dev,				  unsigned long status){	int next_blk, reg = 0;	spin_lock(&dev->slock);	if (UNSET == dev->dmasound.dma_blk) {		dprintk("irq: recording stopped\n");		goto done;	}	if (0 != (status & 0x0f000000))		dprintk("irq: lost %ld\n", (status >> 24) & 0x0f);	if (0 == (status & 0x10000000)) {		/* odd */		if (0 == (dev->dmasound.dma_blk & 0x01))			reg = SAA7134_RS_BA1(6);	} else {		/* even */		if (1 == (dev->dmasound.dma_blk & 0x01))			reg = SAA7134_RS_BA2(6);	}	if (0 == reg) {		dprintk("irq: field oops [%s]\n",			(status & 0x10000000) ? "even" : "odd");		goto done;	}	if (dev->dmasound.read_count >= dev->dmasound.blksize * (dev->dmasound.blocks-2)) {		dprintk("irq: overrun [full=%d/%d] - Blocks in %d\n",dev->dmasound.read_count,			dev->dmasound.bufsize, dev->dmasound.blocks);		spin_unlock(&dev->slock);		snd_pcm_stop(dev->dmasound.substream,SNDRV_PCM_STATE_XRUN);		return;	}	/* next block addr */	next_blk = (dev->dmasound.dma_blk + 2) % dev->dmasound.blocks;	saa_writel(reg,next_blk * dev->dmasound.blksize);	if (debug > 2)		dprintk("irq: ok, %s, next_blk=%d, addr=%x, blocks=%u, size=%u, read=%u\n",			(status & 0x10000000) ? "even" : "odd ", next_blk,			next_blk * dev->dmasound.blksize, dev->dmasound.blocks, dev->dmasound.blksize, dev->dmasound.read_count);	/* update status & wake waiting readers */	dev->dmasound.dma_blk = (dev->dmasound.dma_blk + 1) % dev->dmasound.blocks;	dev->dmasound.read_count += dev->dmasound.blksize;	dev->dmasound.recording_on = reg;	if (dev->dmasound.read_count >= snd_pcm_lib_period_bytes(dev->dmasound.substream)) {		spin_unlock(&dev->slock);		snd_pcm_period_elapsed(dev->dmasound.substream);		spin_lock(&dev->slock);	} done:	spin_unlock(&dev->slock);}/* * IRQ request handler * *   Runs along with saa7134's IRQ handler, discards anything that isn't *   DMA sound * */static irqreturn_t saa7134_alsa_irq(int irq, void *dev_id, struct pt_regs *regs){	struct saa7134_dmasound *dmasound = dev_id;	struct saa7134_dev *dev = dmasound->priv_data;	unsigned long report, status;	int loop, handled = 0;	for (loop = 0; loop < 10; loop++) {		report = saa_readl(SAA7134_IRQ_REPORT);		status = saa_readl(SAA7134_IRQ_STATUS);		if (report & SAA7134_IRQ_REPORT_DONE_RA3) {			handled = 1;			saa_writel(SAA7134_IRQ_REPORT,report);			saa7134_irq_alsa_done(dev, status);		} else {			goto out;		}	}	if (loop == 10) {		dprintk("error! looping IRQ!");	}out:	return IRQ_RETVAL(handled);}/* * ALSA capture trigger * *   - One of the ALSA capture callbacks. * *   Called whenever a capture is started or stopped. Must be defined, *   but there's nothing we want to do here * */static int snd_card_saa7134_capture_trigger(snd_pcm_substream_t * substream,					  int cmd){	snd_pcm_runtime_t *runtime = substream->runtime;	snd_card_saa7134_pcm_t *pcm = runtime->private_data;	struct saa7134_dev *dev=pcm->dev;	int err = 0;	spin_lock(&dev->slock);	if (cmd == SNDRV_PCM_TRIGGER_START) {		/* start dma */		saa7134_dma_start(dev);	} else if (cmd == SNDRV_PCM_TRIGGER_STOP) {		/* stop dma */		saa7134_dma_stop(dev);	} else {		err = -EINVAL;	}	spin_unlock(&dev->slock);	return err;}/* * DMA buffer initialization * *   Uses V4L functions to initialize the DMA. Shouldn't be necessary in *  ALSA, but I was unable to use ALSA's own DMA, and had to force the *  usage of V4L's * *   - Copied verbatim from saa7134-oss. * */static int dsp_buffer_init(struct saa7134_dev *dev){	int err;	BUG_ON(!dev->dmasound.bufsize);	videobuf_dma_init(&dev->dmasound.dma);	err = videobuf_dma_init_kernel(&dev->dmasound.dma, PCI_DMA_FROMDEVICE,				       (dev->dmasound.bufsize + PAGE_SIZE) >> PAGE_SHIFT);	if (0 != err)		return err;	return 0;}/* * DMA buffer release * *   Called after closing the device, during snd_card_saa7134_capture_close * */static int dsp_buffer_free(struct saa7134_dev *dev){	if (!dev->dmasound.blksize)		BUG();	videobuf_dma_free(&dev->dmasound.dma);	dev->dmasound.blocks  = 0;	dev->dmasound.blksize = 0;	dev->dmasound.bufsize = 0;       return 0;}/* * ALSA PCM preparation * *   - One of the ALSA capture callbacks. * *   Called right after the capture device is opened, this function configures *  the buffer using the previously defined functions, allocates the memory, *  sets up the hardware registers, and then starts the DMA. When this function *  returns, the audio should be flowing. * */static int snd_card_saa7134_capture_prepare(snd_pcm_substream_t * substream){	snd_pcm_runtime_t *runtime = substream->runtime;	int bswap, sign;	u32 fmt, control;	snd_card_saa7134_t *saa7134 = snd_pcm_substream_chip(substream);	struct saa7134_dev *dev;	snd_card_saa7134_pcm_t *pcm = runtime->private_data;	pcm->dev->dmasound.substream = substream;	dev = saa7134->dev;	if (snd_pcm_format_width(runtime->format) == 8)		fmt = 0x00;	else		fmt = 0x01;	if (snd_pcm_format_signed(runtime->format))		sign = 1;	else		sign = 0;	if (snd_pcm_format_big_endian(runtime->format))		bswap = 1;	else		bswap = 0;	switch (dev->pci->device) {	  case PCI_DEVICE_ID_PHILIPS_SAA7134:		if (1 == runtime->channels)			fmt |= (1 << 3);		if (2 == runtime->channels)			fmt |= (3 << 3);		if (sign)			fmt |= 0x04;		fmt |= (MIXER_ADDR_TVTUNER == dev->dmasound.input) ? 0xc0 : 0x80;		saa_writeb(SAA7134_NUM_SAMPLES0, ((dev->dmasound.blksize - 1) & 0x0000ff));		saa_writeb(SAA7134_NUM_SAMPLES1, ((dev->dmasound.blksize - 1) & 0x00ff00) >>  8);		saa_writeb(SAA7134_NUM_SAMPLES2, ((dev->dmasound.blksize - 1) & 0xff0000) >> 16);		saa_writeb(SAA7134_AUDIO_FORMAT_CTRL, fmt);		break;	  case PCI_DEVICE_ID_PHILIPS_SAA7133:	  case PCI_DEVICE_ID_PHILIPS_SAA7135:		if (1 == runtime->channels)			fmt |= (1 << 4);		if (2 == runtime->channels)			fmt |= (2 << 4);		if (!sign)			fmt |= 0x04;		saa_writel(SAA7133_NUM_SAMPLES, dev->dmasound.blksize -1);		saa_writel(SAA7133_AUDIO_CHANNEL, 0x543210 | (fmt << 24));		break;	}	dprintk("rec_start: afmt=%d ch=%d  =>  fmt=0x%x swap=%c\n",		runtime->format, runtime->channels, fmt,		bswap ? 'b' : '-');	/* dma: setup channel 6 (= AUDIO) */	control = SAA7134_RS_CONTROL_BURST_16 |		SAA7134_RS_CONTROL_ME |		(dev->dmasound.pt.dma >> 12);	if (bswap)		control |= SAA7134_RS_CONTROL_BSWAP;	saa_writel(SAA7134_RS_BA1(6),0);	saa_writel(SAA7134_RS_BA2(6),dev->dmasound.blksize);	saa_writel(SAA7134_RS_PITCH(6),0);	saa_writel(SAA7134_RS_CONTROL(6),control);	dev->dmasound.rate = runtime->rate;	return 0;}/* * ALSA pointer fetching * *   - One of the ALSA capture callbacks. * *   Called whenever a period elapses, it must return the current hardware *  position of the buffer. *   Also resets the read counter used to prevent overruns * */static snd_pcm_uframes_t snd_card_saa7134_capture_pointer(snd_pcm_substream_t * substream){	snd_pcm_runtime_t *runtime = substream->runtime;	snd_card_saa7134_pcm_t *pcm = runtime->private_data;	struct saa7134_dev *dev=pcm->dev;	if (dev->dmasound.read_count) {		dev->dmasound.read_count  -= snd_pcm_lib_period_bytes(substream);		dev->dmasound.read_offset += snd_pcm_lib_period_bytes(substream);		if (dev->dmasound.read_offset == dev->dmasound.bufsize)			dev->dmasound.read_offset = 0;	}	return bytes_to_frames(runtime, dev->dmasound.read_offset);}/* * ALSA hardware capabilities definition */static snd_pcm_hardware_t snd_card_saa7134_capture ={	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |				 SNDRV_PCM_INFO_BLOCK_TRANSFER |				 SNDRV_PCM_INFO_MMAP_VALID),	.formats =		SNDRV_PCM_FMTBIT_S16_LE | \				SNDRV_PCM_FMTBIT_S16_BE | \				SNDRV_PCM_FMTBIT_S8 | \				SNDRV_PCM_FMTBIT_U8 | \				SNDRV_PCM_FMTBIT_U16_LE | \				SNDRV_PCM_FMTBIT_U16_BE,	.rates =		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000,	.rate_min =		32000,	.rate_max =		48000,	.channels_min =		1,	.channels_max =		2,	.buffer_bytes_max =	(256*1024),	.period_bytes_min =	64,	.period_bytes_max =	(256*1024),	.periods_min =		2,	.periods_max =		1024,};static void snd_card_saa7134_runtime_free(snd_pcm_runtime_t *runtime){	snd_card_saa7134_pcm_t *pcm = runtime->private_data;	kfree(pcm);}/* * ALSA hardware params * *   - One of the ALSA capture callbacks. * *   Called on initialization, right before the PCM preparation * */static int snd_card_saa7134_hw_params(snd_pcm_substream_t * substream,				    snd_pcm_hw_params_t * hw_params){	snd_card_saa7134_t *saa7134 = snd_pcm_substream_chip(substream);	struct saa7134_dev *dev;	unsigned int period_size, periods;	int err;	period_size = params_period_bytes(hw_params);	periods = params_periods(hw_params);	snd_assert(period_size >= 0x100 && period_size <= 0x10000,		   return -EINVAL);	snd_assert(periods >= 2, return -EINVAL);	snd_assert(period_size * periods <= 1024 * 1024, return -EINVAL);	dev = saa7134->dev;	if (dev->dmasound.blocks == periods &&	    dev->dmasound.blksize == period_size)		return 0;	/* release the old buffer */	if (substream->runtime->dma_area) {		saa7134_pgtable_free(dev->pci, &dev->dmasound.pt);		videobuf_dma_pci_unmap(dev->pci, &dev->dmasound.dma);		dsp_buffer_free(dev);		substream->runtime->dma_area = NULL;	}	dev->dmasound.blocks  = periods;	dev->dmasound.blksize = period_size;	dev->dmasound.bufsize = period_size * periods;	err = dsp_buffer_init(dev);	if (0 != err) {		dev->dmasound.blocks  = 0;		dev->dmasound.blksize = 0;		dev->dmasound.bufsize = 0;		return err;	}