Linux-2.6.12-rc2
Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.
Let it rip!
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
new file mode 100644
index 0000000..174bc03
--- /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:
+ */