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authorRussell King <rmk@dyn-67.arm.linux.org.uk>2009-03-28 20:29:51 +0000
committerRussell King <rmk+kernel@arm.linux.org.uk>2009-03-28 20:29:51 +0000
commited40d0c472b136682b2fcba05f89762859c7374f (patch)
tree076b83a26bcd63d6158463735dd34c10bbc591dc /sound/arm
parent9e495834e59ca9b29f1a1f63b9f5533bb022ac49 (diff)
parent5d80f8e5a9dc9c9a94d4aeaa567e219a808b8a4a (diff)
downloadlinux-2.6.38-lt-ux500-ed40d0c472b136682b2fcba05f89762859c7374f.tar.gz
Merge branch 'origin' into devel
Conflicts: sound/soc/pxa/pxa2xx-i2s.c
Diffstat (limited to 'sound/arm')
-rw-r--r--sound/arm/Kconfig11
-rw-r--r--sound/arm/Makefile3
-rw-r--r--sound/arm/aaci.c7
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c71
-rw-r--r--sound/arm/pxa2xx-ac97.c7
-rw-r--r--sound/arm/sa11xx-uda1341.c983
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:
- */