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Diffstat (limited to 'sound/arm/sa11xx-uda1341.c')
| -rw-r--r-- | sound/arm/sa11xx-uda1341.c | 973 |
1 files changed, 973 insertions, 0 deletions
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c new file mode 100644 index 00000000000..174bc032d1a --- /dev/null +++ b/sound/arm/sa11xx-uda1341.c @@ -0,0 +1,973 @@ +/* + * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard + * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License. + * + * History: + * + * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS + * 2002-03-20 Tomas Kasparek playback over ALSA is working + * 2002-03-28 Tomas Kasparek playback over OSS emulation is working + * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA) + * 2002-03-29 Tomas Kasparek capture is working (OSS emulation) + * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates) + * 2003-02-14 Brian Avery fixed full duplex mode, other updates + * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL) + * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel + * working suspend and resume + * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again + * merged HAL layer (patches from Brian) + */ + +/* $Id: sa11xx-uda1341.c,v 1.21 2005/01/28 19:34:04 tiwai Exp $ */ + +/*************************************************************************************************** +* +* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai +* available in the Alsa doc section on the website +* +* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100. +* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated +* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it. +* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the +* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which +* is a mem loc that always decodes to 0's w/ no off chip access. +* +* Some alsa terminology: +* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes +* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte +* buffer and 4 periods in the runtime structure this means we'll get an int every 256 +* bytes or 4 times per buffer. +* A number of the sizes are in frames rather than bytes, use frames_to_bytes and +* bytes_to_frames to convert. The easiest way to tell the units is to look at the +* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t +* +* Notes about the pointer fxn: +* The pointer fxn needs to return the offset into the dma buffer in frames. +* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts. +* +* Notes about pause/resume +* Implementing this would be complicated so it's skipped. The problem case is: +* A full duplex connection is going, then play is paused. At this point you need to start xmitting +* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd +* need to save off the dma info, and restore it properly on a resume. Yeach! +* +* Notes about transfer methods: +* The async write calls fail. I probably need to implement something else to support them? +* +***************************************************************************************************/ + +#include <linux/config.h> +#include <sound/driver.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/errno.h> +#include <linux/ioctl.h> +#include <linux/delay.h> +#include <linux/slab.h> + +#ifdef CONFIG_PM +#include <linux/pm.h> +#endif + +#include <asm/hardware.h> +#include <asm/arch/h3600.h> +#include <asm/mach-types.h> +#include <asm/dma.h> + +#ifdef CONFIG_H3600_HAL +#include <asm/semaphore.h> +#include <asm/uaccess.h> +#include <asm/arch/h3600_hal.h> +#endif + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +#include <linux/l3/l3.h> + +#undef DEBUG_MODE +#undef DEBUG_FUNCTION_NAMES +#include <sound/uda1341.h> + +/* + * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? + * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this + * module for Familiar 0.6.1 + */ +#ifdef CONFIG_H3600_HAL +#define HH_VERSION 1 +#endif + +/* {{{ Type definitions */ + +MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); +MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); + +static char *id = NULL; /* ID for this card */ + +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); + +typedef struct audio_stream { + char *id; /* identification string */ + int stream_id; /* numeric identification */ + dma_device_t dma_dev; /* device identifier for DMA */ +#ifdef HH_VERSION + dmach_t dmach; /* dma channel identification */ +#else + dma_regs_t *dma_regs; /* points to our DMA registers */ +#endif + int active:1; /* we are using this stream for transfer now */ + int period; /* current transfer period */ + int periods; /* current count of periods registerd in the DMA engine */ + int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ + unsigned int old_offset; + spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ + snd_pcm_substream_t *stream; +}audio_stream_t; + +typedef struct snd_card_sa11xx_uda1341 { + snd_card_t *card; + struct l3_client *uda1341; + snd_pcm_t *pcm; + long samplerate; + audio_stream_t s[2]; /* playback & capture */ +} sa11xx_uda1341_t; + +static struct snd_card_sa11xx_uda1341 *sa11xx_uda1341 = NULL; + +static unsigned int rates[] = { + 8000, 10666, 10985, 14647, + 16000, 21970, 22050, 24000, + 29400, 32000, 44100, 48000, +}; + +static snd_pcm_hw_constraint_list_t hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +/* }}} */ + +/* {{{ Clock and sample rate stuff */ + +/* + * Stop-gap solution until rest of hh.org HAL stuff is merged. + */ +#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) +#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) + +#ifdef CONFIG_SA1100_H3XXX +#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) +#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) +#else +#error This driver could serve H3x00 handhelds only! +#endif + +static void sa11xx_uda1341_set_audio_clock(long val) +{ + switch (val) { + case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ + GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; + break; + + case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ + GPSR = GPIO_H3600_CLK_SET0; + GPCR = GPIO_H3600_CLK_SET1; + break; + + case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ + GPCR = GPIO_H3600_CLK_SET0; + GPSR = GPIO_H3600_CLK_SET1; + break; + + case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ + GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; + break; + } +} + +static void sa11xx_uda1341_set_samplerate(sa11xx_uda1341_t *sa11xx_uda1341, long rate) +{ + int clk_div = 0; + int clk=0; + + /* We don't want to mess with clocks when frames are in flight */ + Ser4SSCR0 &= ~SSCR0_SSE; + /* wait for any frame to complete */ + udelay(125); + + /* + * We have the following clock sources: + * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz + * Those can be divided either by 256, 384 or 512. + * This makes up 12 combinations for the following samplerates... + */ + if (rate >= 48000) + rate = 48000; + else if (rate >= 44100) + rate = 44100; + else if (rate >= 32000) + rate = 32000; + else if (rate >= 29400) + rate = 29400; + else if (rate >= 24000) + rate = 24000; + else if (rate >= 22050) + rate = 22050; + else if (rate >= 21970) + rate = 21970; + else if (rate >= 16000) + rate = 16000; + else if (rate >= 14647) + rate = 14647; + else if (rate >= 10985) + rate = 10985; + else if (rate >= 10666) + rate = 10666; + else + rate = 8000; + + /* Set the external clock generator */ +#ifdef CONFIG_H3600_HAL + h3600_audio_clock(rate); +#else + sa11xx_uda1341_set_audio_clock(rate); +#endif + + /* Select the clock divisor */ + switch (rate) { + case 8000: + case 10985: + case 22050: + case 24000: + clk = F512; + clk_div = SSCR0_SerClkDiv(16); + break; + case 16000: + case 21970: + case 44100: + case 48000: + clk = F256; + clk_div = SSCR0_SerClkDiv(8); + break; + case 10666: + case 14647: + case 29400: + case 32000: + clk = F384; + clk_div = SSCR0_SerClkDiv(12); + break; + } + + /* FMT setting should be moved away when other FMTs are added (FIXME) */ + l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); + + l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); + Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; + sa11xx_uda1341->samplerate = rate; +} + +/* }}} */ + +/* {{{ HW init and shutdown */ + +static void sa11xx_uda1341_audio_init(sa11xx_uda1341_t *sa11xx_uda1341) +{ + unsigned long flags; + + /* Setup DMA stuff */ + sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; + sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; + sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; + + sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; + sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; + sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; + + /* Initialize the UDA1341 internal state */ + + /* Setup the uarts */ + local_irq_save(flags); + GAFR |= (GPIO_SSP_CLK); + GPDR &= ~(GPIO_SSP_CLK); + Ser4SSCR0 = 0; + Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); + Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; + Ser4SSCR0 |= SSCR0_SSE; + local_irq_restore(flags); + + /* Enable the audio power */ +#ifdef CONFIG_H3600_HAL + h3600_audio_power(AUDIO_RATE_DEFAULT); +#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); + set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); + set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); +#endif + + /* Wait for the UDA1341 to wake up */ + mdelay(1); //FIXME - was removed by Perex - Why? + + /* Initialize the UDA1341 internal state */ + l3_open(sa11xx_uda1341->uda1341); + + /* external clock configuration (after l3_open - regs must be initialized */ + sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); + + /* Wait for the UDA1341 to wake up */ + set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); + mdelay(1); + + /* make the left and right channels unswapped (flip the WS latch) */ + Ser4SSDR = 0; + +#ifdef CONFIG_H3600_HAL + h3600_audio_mute(0); +#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); +#endif +} + +static void sa11xx_uda1341_audio_shutdown(sa11xx_uda1341_t *sa11xx_uda1341) +{ + /* mute on */ +#ifdef CONFIG_H3600_HAL + h3600_audio_mute(1); +#else + set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); +#endif + + /* disable the audio power and all signals leading to the audio chip */ + l3_close(sa11xx_uda1341->uda1341); + Ser4SSCR0 = 0; + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); + + /* power off and mute off */ + /* FIXME - is muting off necesary??? */ +#ifdef CONFIG_H3600_HAL + h3600_audio_power(0); + h3600_audio_mute(0); +#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); +#endif +} + +/* }}} */ + +/* {{{ DMA staff */ + +/* + * these are the address and sizes used to fill the xmit buffer + * so we can get a clock in record only mode + */ +#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS +#define FORCE_CLOCK_SIZE 4096 // was 2048 + +// FIXME Why this value exactly - wrote comment +#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ + +#ifdef HH_VERSION + +static int audio_dma_request(audio_stream_t *s, void (*callback)(void *, int)) +{ + int ret; + + ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); + if (ret < 0) { + printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); + return ret; + } + sa1100_dma_set_callback(s->dmach, callback); + return 0; +} + +static inline void audio_dma_free(audio_stream_t *s) +{ + sa1100_free_dma(s->dmach); + s->dmach = -1; +} + +#else + +static int audio_dma_request(audio_stream_t *s, void (*callback)(void *)) +{ + int ret; + + ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); + if (ret < 0) + printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); + return ret; +} + +static void audio_dma_free(audio_stream_t *s) +{ + sa1100_free_dma((s)->dma_regs); + (s)->dma_regs = 0; +} + +#endif + +static u_int audio_get_dma_pos(audio_stream_t *s) +{ + snd_pcm_substream_t * substream = s->stream; + snd_pcm_runtime_t *runtime = substream->runtime; + unsigned int offset; + unsigned long flags; + dma_addr_t addr; + + // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel + spin_lock_irqsave(&s->dma_lock, flags); +#ifdef HH_VERSION + sa1100_dma_get_current(s->dmach, NULL, &addr); +#else + addr = sa1100_get_dma_pos((s)->dma_regs); +#endif + offset = addr - runtime->dma_addr; + spin_unlock_irqrestore(&s->dma_lock, flags); + + offset = bytes_to_frames(runtime,offset); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +/* + * this stops the dma and clears the dma ptrs + */ +static void audio_stop_dma(audio_stream_t *s) +{ + unsigned long flags; + + spin_lock_irqsave(&s->dma_lock, flags); + s->active = 0; + s->period = 0; + /* this stops the dma channel and clears the buffer ptrs */ +#ifdef HH_VERSION + sa1100_dma_flush_all(s->dmach); +#else + sa1100_clear_dma(s->dma_regs); +#endif + spin_unlock_irqrestore(&s->dma_lock, flags); +} + +static void audio_process_dma(audio_stream_t *s) +{ + snd_pcm_substream_t *substream = s->stream; + snd_pcm_runtime_t *runtime; + unsigned int dma_size; + unsigned int offset; + int ret; + + /* we are requested to process synchronization DMA transfer */ + if (s->tx_spin) { + snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return); + /* fill the xmit dma buffers and return */ +#ifdef HH_VERSION + sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); +#else + while (1) { + ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); + if (ret) + return; + } +#endif + return; + } + + /* must be set here - only valid for running streams, not for forced_clock dma fills */ + runtime = substream->runtime; + while (s->active && s->periods < runtime->periods) { + dma_size = frames_to_bytes(runtime, runtime->period_size); + if (s->old_offset) { + /* a little trick, we need resume from old position */ + offset = frames_to_bytes(runtime, s->old_offset - 1); + s->old_offset = 0; + s->periods = 0; + s->period = offset / dma_size; + offset %= dma_size; + dma_size = dma_size - offset; + if (!dma_size) + continue; /* special case */ + } else { + offset = dma_size * s->period; + snd_assert(dma_size <= DMA_BUF_SIZE, ); + } +#ifdef HH_VERSION + ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size); + if (ret) + return; //FIXME +#else + ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size); + if (ret) { + printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret); + return; + } +#endif + + s->period++; + s->period %= runtime->periods; + s->periods++; + } +} + +#ifdef HH_VERSION +static void audio_dma_callback(void *data, int size) +#else +static void audio_dma_callback(void *data) +#endif +{ + audio_stream_t *s = data; + + /* + * If we are getting a callback for an active stream then we inform + * the PCM middle layer we've finished a period + */ + if (s->active) + snd_pcm_period_elapsed(s->stream); + + spin_lock(&s->dma_lock); + if (!s->tx_spin && s->periods > 0) + s->periods--; + audio_process_dma(s); + spin_unlock(&s->dma_lock); +} + +/* }}} */ + +/* {{{ PCM setting */ + +/* {{{ trigger & timer */ + +static int snd_sa11xx_uda1341_trigger(snd_pcm_substream_t * substream, int cmd) +{ + sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream); + int stream_id = substream->pstr->stream; + audio_stream_t *s = &chip->s[stream_id]; + audio_stream_t *s1 = &chip->s[stream_id ^ 1]; + int err = 0; + + /* note local interrupts are already disabled in the midlevel code */ + spin_lock(&s->dma_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* now we need to make sure a record only stream has a clock */ + if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { + /* we need to force fill the xmit DMA with zeros */ + s1->tx_spin = 1; + audio_process_dma(s1); + } + /* this case is when you were recording then you turn on a + * playback stream so we stop (also clears it) the dma first, + * clear the sync flag and then we let it turned on + */ + else { + s->tx_spin = 0; + } + + /* requested stream startup */ + s->active = 1; + audio_process_dma(s); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* requested stream shutdown */ + audio_stop_dma(s); + + /* + * now we need to make sure a record only stream has a clock + * so if we're stopping a playback with an active capture + * we need to turn the 0 fill dma on for the xmit side + */ + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) { + /* we need to force fill the xmit DMA with zeros */ + s->tx_spin = 1; + audio_process_dma(s); + } + /* + * we killed a capture only stream, so we should also kill + * the zero fill transmit + */ + else { + if (s1->tx_spin) { + s1->tx_spin = 0; + audio_stop_dma(s1); + } + } + + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + s->active = 0; +#ifdef HH_VERSION + sa1100_dma_stop(s->dmach); +#else + //FIXME - DMA API +#endif + s->old_offset = audio_get_dma_pos(s) + 1; +#ifdef HH_VERSION + sa1100_dma_flush_all(s->dmach); +#else + //FIXME - DMA API +#endif + s->periods = 0; + break; + case SNDRV_PCM_TRIGGER_RESUME: + s->active = 1; + s->tx_spin = 0; + audio_process_dma(s); + if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { + s1->tx_spin = 1; + audio_process_dma(s1); + } + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: +#ifdef HH_VERSION + sa1100_dma_stop(s->dmach); +#else + //FIXME - DMA API +#endif + s->active = 0; + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) { + if (s1->active) { + s->tx_spin = 1; + s->old_offset = audio_get_dma_pos(s) + 1; +#ifdef HH_VERSION + sa1100_dma_flush_all(s->dmach); +#else + //FIXME - DMA API +#endif + audio_process_dma(s); + } + } else { + if (s1->tx_spin) { + s1->tx_spin = 0; +#ifdef HH_VERSION + sa1100_dma_flush_all(s1->dmach); +#else + //FIXME - DMA API +#endif + } + } + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + s->active = 1; + if (s->old_offset) { + s->tx_spin = 0; + audio_process_dma(s); + break; + } + if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { + s1->tx_spin = 1; + audio_process_dma(s1); + } +#ifdef HH_VERSION + sa1100_dma_resume(s->dmach); +#else + //FIXME - DMA API +#endif + break; + default: + err = -EINVAL; + break; + } + spin_unlock(&s->dma_lock); + return err; +} + +static int snd_sa11xx_uda1341_prepare(snd_pcm_substream_t * substream) +{ + sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + audio_stream_t *s = &chip->s[substream->pstr->stream]; + + /* set requested samplerate */ + sa11xx_uda1341_set_samplerate(chip, runtime->rate); + + /* set requestd format when available */ + /* set FMT here !!! FIXME */ + + s->period = 0; + s->periods = 0; + + return 0; +} + +static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(snd_pcm_substream_t * substream) +{ + sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream); + return audio_get_dma_pos(&chip->s[substream->pstr->stream]); +} + +/* }}} */ + +static snd_pcm_hardware_t snd_sa11xx_uda1341_capture = +{ + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64*1024, + .period_bytes_min = 64, + .period_bytes_max = DMA_BUF_SIZE, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static snd_pcm_hardware_t snd_sa11xx_uda1341_playback = +{ + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64*1024, + .period_bytes_min = 64, + .period_bytes_max = DMA_BUF_SIZE, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_card_sa11xx_uda1341_open(snd_pcm_substream_t * substream) +{ + sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + int stream_id = substream->pstr->stream; + int err; + + chip->s[stream_id].stream = substream; + + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = snd_sa11xx_uda1341_playback; + else + runtime->hw = snd_sa11xx_uda1341_capture; + if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + return err; + if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) + return err; + + return 0; +} + +static int snd_card_sa11xx_uda1341_close(snd_pcm_substream_t * substream) +{ + sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream); + + chip->s[substream->pstr->stream].stream = NULL; + return 0; +} + +/* {{{ HW params & free */ + +static int snd_sa11xx_uda1341_hw_params(snd_pcm_substream_t * substream, + snd_pcm_hw_params_t * hw_params) +{ + + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_sa11xx_uda1341_hw_free(snd_pcm_substream_t * substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* }}} */ + +static snd_pcm_ops_t snd_card_sa11xx_uda1341_playback_ops = { + .open = snd_card_sa11xx_uda1341_open, + .close = snd_card_sa11xx_uda1341_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sa11xx_uda1341_hw_params, + .hw_free = snd_sa11xx_uda1341_hw_free, + .prepare = snd_sa11xx_uda1341_prepare, + .trigger = snd_sa11xx_uda1341_trigger, + .pointer = snd_sa11xx_uda1341_pointer, +}; + +static snd_pcm_ops_t snd_card_sa11xx_uda1341_capture_ops = { + .open = snd_card_sa11xx_uda1341_open, + .close = snd_card_sa11xx_uda1341_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sa11xx_uda1341_hw_params, + .hw_free = snd_sa11xx_uda1341_hw_free, + .prepare = snd_sa11xx_uda1341_prepare, + .trigger = snd_sa11xx_uda1341_trigger, + .pointer = snd_sa11xx_uda1341_pointer, +}; + +static int __init snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341_t *sa11xx_uda1341, int device) +{ + snd_pcm_t *pcm; + int err; + + if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0) + return err; + + /* + * this sets up our initial buffers and sets the dma_type to isa. + * isa works but I'm not sure why (or if) it's the right choice + * this may be too large, trying it for now + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_ISA, + snd_pcm_dma_flags(0), + 64*1024, 64*1024); + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops); + pcm->private_data = sa11xx_uda1341; + pcm->info_flags = 0; + strcpy(pcm->name, "UDA1341 PCM"); + + sa11xx_uda1341_audio_init(sa11xx_uda1341); + + /* setup DMA controller */ + audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback); + audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback); + + sa11xx_uda1341->pcm = pcm; + + return 0; +} + +/* }}} */ + +/* {{{ module init & exit */ + +#ifdef CONFIG_PM + +static int snd_sa11xx_uda1341_suspend(snd_card_t *card, pm_message_t state) +{ + sa11xx_uda1341_t *chip = card->pm_private_data; + + snd_pcm_suspend_all(chip->pcm); +#ifdef HH_VERSION + sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); + sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); +#else + //FIXME +#endif + l3_command(chip->uda1341, CMD_SUSPEND, NULL); + sa11xx_uda1341_audio_shutdown(chip); + return 0; +} + +static int snd_sa11xx_uda1341_resume(snd_card_t *card) +{ + sa11xx_uda1341_t *chip = card->pm_private_data; + + sa11xx_uda1341_audio_init(chip); + l3_command(chip->uda1341, CMD_RESUME, NULL); +#ifdef HH_VERSION + sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); + sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); +#else + //FIXME +#endif + return 0; +} +#endif /* COMFIG_PM */ + +void snd_sa11xx_uda1341_free(snd_card_t *card) +{ + sa11xx_uda1341_t *chip = card->private_data; + + audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); + audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); + sa11xx_uda1341 = NULL; + card->private_data = NULL; + kfree(chip); +} + +static int __init sa11xx_uda1341_init(void) +{ + int err; + snd_card_t *card; + + if (!machine_is_h3xxx()) + return -ENODEV; + + /* register the soundcard */ + card = snd_card_new(-1, id, THIS_MODULE, sizeof(sa11xx_uda1341_t)); + if (card == NULL) + return -ENOMEM; + + sa11xx_uda1341 = kcalloc(1, sizeof(*sa11xx_uda1341), GFP_KERNEL); + if (sa11xx_uda1341 == NULL) + return -ENOMEM; + spin_lock_init(&chip->s[0].dma_lock); + spin_lock_init(&chip->s[1].dma_lock); + + card->private_data = (void *)sa11xx_uda1341; + card->private_free = snd_sa11xx_uda1341_free; + + sa11xx_uda1341->card = card; + sa11xx_uda1341->samplerate = AUDIO_RATE_DEFAULT; + + // mixer + if ((err = snd_chip_uda1341_mixer_new(sa11xx_uda1341->card, &sa11xx_uda1341->uda1341))) + goto nodev; + + // PCM + if ((err = snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341, 0)) < 0) + goto nodev; + + snd_card_set_generic_pm_callback(card, + snd_sa11xx_uda1341_suspend, snd_sa11_uda1341_resume, + sa11xx_uda1341); + + strcpy(card->driver, "UDA1341"); + strcpy(card->shortname, "H3600 UDA1341TS"); + sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); + + if ((err = snd_card_register(card)) == 0) { + printk( KERN_INFO "iPAQ audio support initialized\n" ); + return 0; + } + + nodev: + snd_card_free(card); + return err; +} + +static void __exit sa11xx_uda1341_exit(void) +{ + snd_card_free(sa11xx_uda1341->card); +} + +module_init(sa11xx_uda1341_init); +module_exit(sa11xx_uda1341_exit); + +/* }}} */ + +/* + * Local variables: + * indent-tabs-mode: t + * End: + */ |