diff options
Diffstat (limited to 'sound/arm')
| -rw-r--r-- | sound/arm/Kconfig | 11 | ||||
| -rw-r--r-- | sound/arm/Makefile | 3 | ||||
| -rw-r--r-- | sound/arm/aaci.c | 7 | ||||
| -rw-r--r-- | sound/arm/pxa2xx-ac97-lib.c | 71 | ||||
| -rw-r--r-- | sound/arm/pxa2xx-ac97.c | 7 | ||||
| -rw-r--r-- | sound/arm/sa11xx-uda1341.c | 983 |
6 files changed, 73 insertions, 1009 deletions
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de48d81..885683a3b0b 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -11,17 +11,6 @@ menuconfig SND_ARM if SND_ARM -config SND_SA11XX_UDA1341 - tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && L3 - select SND_PCM - help - Say Y here if you have a Compaq iPaq H3x00 handheld computer - and want to use its Philips UDA 1341 audio chip. - - To compile this driver as a module, choose M here: the module - will be called snd-sa11xx-uda1341. - config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de11de8..5a549ed6c8a 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -2,9 +2,6 @@ # Makefile for ALSA # -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o -snd-sa11xx-uda1341-objs := sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o devdma.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 772901e41ec..7fbd68fab94 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -995,10 +995,11 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) { struct aaci *aaci; struct snd_card *card; + int err; - card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, - THIS_MODULE, sizeof(struct aaci)); - if (card == NULL) + err = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct aaci), &card); + if (err < 0) return NULL; card->private_free = aaci_free_card; diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 718d06640dd..7793d2a511c 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -30,6 +30,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int reset_gpio; /* * Beware PXA27x bugs: @@ -41,6 +42,45 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -136,10 +176,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -307,8 +347,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -319,6 +359,27 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; + struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + + if (pdata) { + switch (pdata->reset_gpio) { + case 95: + case 113: + reset_gpio = pdata->reset_gpio; + break; + case 0: + reset_gpio = 113; + break; + case -1: + break; + default: + dev_err(&dev->dev, "Invalid reset GPIO %d\n", + pdata->reset_gpio); + } + } else { + if (cpu_is_pxa27x()) + reset_gpio = 113; + } if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); @@ -329,7 +390,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ccec48fc8e3..c570ebd9d17 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -171,10 +171,9 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) struct snd_ac97_template ac97_template; int ret; - ret = -ENOMEM; - card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, - THIS_MODULE, 0); - if (!card) + ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (ret < 0) goto err; card->dev = &dev->dev; diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c deleted file mode 100644 index 1dcd51d81d1..00000000000 --- a/sound/arm/sa11xx-uda1341.c +++ /dev/null @@ -1,983 +0,0 @@ -/* - * 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) - */ - -/*************************************************************************************************** -* -* 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/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/err.h> -#include <linux/platform_device.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 <mach/hardware.h> -#include <mach/h3600.h> -#include <asm/mach-types.h> -#include <asm/dma.h> - -#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 - */ - -/* {{{ 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; /* ID for this card */ - -module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); - -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 - unsigned 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) */ - struct snd_pcm_substream *stream; -}; - -struct sa11xx_uda1341 { - struct snd_card *card; - struct l3_client *uda1341; - struct snd_pcm *pcm; - long samplerate; - struct audio_stream s[2]; /* playback & capture */ -}; - -static unsigned int rates[] = { - 8000, 10666, 10985, 14647, - 16000, 21970, 22050, 24000, - 29400, 32000, 44100, 48000, -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct platform_device *device; - -/* }}} */ - -/* {{{ 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(struct sa11xx_uda1341 *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 */ - - sa11xx_uda1341_set_audio_clock(rate); - - /* 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(struct sa11xx_uda1341 *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 */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* 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; - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - /* mute on */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* 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??? */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -/* }}} */ - -/* {{{ 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(struct audio_stream *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(struct audio_stream *s) -{ - sa1100_free_dma(s->dmach); - s->dmach = -1; -} - -#else - -static int audio_dma_request(struct audio_stream *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(struct audio_stream *s) -{ - sa1100_free_dma(s->dma_regs); - s->dma_regs = 0; -} - -#endif - -static u_int audio_get_dma_pos(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *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(struct audio_stream *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(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime; - unsigned int dma_size; - unsigned int offset; - int ret; - - /* we are requested to process synchronization DMA transfer */ - if (s->tx_spin) { - if (snd_BUG_ON(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_BUG_ON(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 -{ - struct audio_stream *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(struct snd_pcm_substream *substream, int cmd) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - int stream_id = substream->pstr->stream; - struct audio_stream *s = &chip->s[stream_id]; - struct audio_stream *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(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audio_stream *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(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - return audio_get_dma_pos(&chip->s[substream->pstr->stream]); -} - -/* }}} */ - -static struct snd_pcm_hardware 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 struct snd_pcm_hardware 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(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *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(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *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(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - -static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -/* }}} */ - -static struct snd_pcm_ops 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 struct snd_pcm_ops 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(struct sa11xx_uda1341 *sa11xx_uda1341, int device) -{ - struct snd_pcm *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_DEV, - snd_dma_isa_data(), - 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(struct platform_device *devptr, - pm_message_t state) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - 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(struct platform_device *devptr) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->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 - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif /* COMFIG_PM */ - -void snd_sa11xx_uda1341_free(struct snd_card *card) -{ - struct sa11xx_uda1341 *chip = card->private_data; - - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); -} - -static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) -{ - int err; - struct snd_card *card; - struct sa11xx_uda1341 *chip; - - /* register the soundcard */ - card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341)); - if (card == NULL) - return -ENOMEM; - - chip = card->private_data; - spin_lock_init(&chip->s[0].dma_lock); - spin_lock_init(&chip->s[1].dma_lock); - - card->private_free = snd_sa11xx_uda1341_free; - chip->card = card; - chip->samplerate = AUDIO_RATE_DEFAULT; - - // mixer - if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341))) - goto nodev; - - // PCM - if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0) - goto nodev; - - strcpy(card->driver, "UDA1341"); - strcpy(card->shortname, "H3600 UDA1341TS"); - sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); - - snd_card_set_dev(card, &devptr->dev); - - if ((err = snd_card_register(card)) == 0) { - printk( KERN_INFO "iPAQ audio support initialized\n" ); - platform_set_drvdata(devptr, card); - return 0; - } - - nodev: - snd_card_free(card); - return err; -} - -static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr) -{ - snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; -} - -#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341" - -static struct platform_driver sa11xx_uda1341_driver = { - .probe = sa11xx_uda1341_probe, - .remove = __devexit_p(sa11xx_uda1341_remove), -#ifdef CONFIG_PM - .suspend = snd_sa11xx_uda1341_suspend, - .resume = snd_sa11xx_uda1341_resume, -#endif - .driver = { - .name = SA11XX_UDA1341_DRIVER, - }, -}; - -static int __init sa11xx_uda1341_init(void) -{ - int err; - - if (!machine_is_h3xxx()) - return -ENODEV; - if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) - return err; - device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (!IS_ERR(device)) { - if (platform_get_drvdata(device)) - return 0; - platform_device_unregister(device); - err = -ENODEV; - } else - err = PTR_ERR(device); - platform_driver_unregister(&sa11xx_uda1341_driver); - return err; -} - -static void __exit sa11xx_uda1341_exit(void) -{ - platform_device_unregister(device); - platform_driver_unregister(&sa11xx_uda1341_driver); -} - -module_init(sa11xx_uda1341_init); -module_exit(sa11xx_uda1341_exit); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ |