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976 lines
26 KiB
976 lines
26 KiB
/*
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* Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
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* Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License.
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*
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* History:
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*
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* 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS
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* 2002-03-20 Tomas Kasparek playback over ALSA is working
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* 2002-03-28 Tomas Kasparek playback over OSS emulation is working
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* 2002-03-29 Tomas Kasparek basic capture is working (native ALSA)
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* 2002-03-29 Tomas Kasparek capture is working (OSS emulation)
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* 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates)
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* 2003-02-14 Brian Avery fixed full duplex mode, other updates
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* 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL)
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* 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
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* working suspend and resume
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* 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again
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* merged HAL layer (patches from Brian)
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*/
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/* $Id: sa11xx-uda1341.c,v 1.23 2005/09/09 13:22:34 tiwai Exp $ */
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/***************************************************************************************************
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*
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* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
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* available in the Alsa doc section on the website
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*
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* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
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* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
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* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
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* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
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* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
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* is a mem loc that always decodes to 0's w/ no off chip access.
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*
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* Some alsa terminology:
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* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes
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* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
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* buffer and 4 periods in the runtime structure this means we'll get an int every 256
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* bytes or 4 times per buffer.
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* A number of the sizes are in frames rather than bytes, use frames_to_bytes and
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* bytes_to_frames to convert. The easiest way to tell the units is to look at the
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* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
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*
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* Notes about the pointer fxn:
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* The pointer fxn needs to return the offset into the dma buffer in frames.
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* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
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*
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* Notes about pause/resume
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* Implementing this would be complicated so it's skipped. The problem case is:
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* A full duplex connection is going, then play is paused. At this point you need to start xmitting
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* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd
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* need to save off the dma info, and restore it properly on a resume. Yeach!
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*
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* Notes about transfer methods:
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* The async write calls fail. I probably need to implement something else to support them?
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*
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***************************************************************************************************/
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#include <linux/config.h>
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#include <sound/driver.h>
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/init.h>
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#include <linux/errno.h>
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#include <linux/ioctl.h>
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#include <linux/delay.h>
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#include <linux/slab.h>
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#ifdef CONFIG_PM
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#include <linux/pm.h>
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#endif
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#include <asm/hardware.h>
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#include <asm/arch/h3600.h>
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#include <asm/mach-types.h>
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#include <asm/dma.h>
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#ifdef CONFIG_H3600_HAL
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#include <asm/semaphore.h>
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#include <asm/uaccess.h>
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#include <asm/arch/h3600_hal.h>
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#endif
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/initval.h>
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#include <linux/l3/l3.h>
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#undef DEBUG_MODE
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#undef DEBUG_FUNCTION_NAMES
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#include <sound/uda1341.h>
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/*
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* FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
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* We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
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* module for Familiar 0.6.1
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*/
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#ifdef CONFIG_H3600_HAL
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#define HH_VERSION 1
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#endif
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/* {{{ Type definitions */
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MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
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MODULE_LICENSE("GPL");
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MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
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MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
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static char *id = NULL; /* ID for this card */
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module_param(id, charp, 0444);
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MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
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typedef struct audio_stream {
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char *id; /* identification string */
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int stream_id; /* numeric identification */
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dma_device_t dma_dev; /* device identifier for DMA */
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#ifdef HH_VERSION
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dmach_t dmach; /* dma channel identification */
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#else
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dma_regs_t *dma_regs; /* points to our DMA registers */
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#endif
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int active:1; /* we are using this stream for transfer now */
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int period; /* current transfer period */
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int periods; /* current count of periods registerd in the DMA engine */
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int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
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unsigned int old_offset;
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spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
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snd_pcm_substream_t *stream;
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}audio_stream_t;
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typedef struct snd_card_sa11xx_uda1341 {
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snd_card_t *card;
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struct l3_client *uda1341;
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snd_pcm_t *pcm;
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long samplerate;
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audio_stream_t s[2]; /* playback & capture */
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} sa11xx_uda1341_t;
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static struct snd_card_sa11xx_uda1341 *sa11xx_uda1341 = NULL;
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static unsigned int rates[] = {
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8000, 10666, 10985, 14647,
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16000, 21970, 22050, 24000,
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29400, 32000, 44100, 48000,
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};
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static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
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.count = ARRAY_SIZE(rates),
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.list = rates,
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.mask = 0,
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};
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/* }}} */
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/* {{{ Clock and sample rate stuff */
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/*
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* Stop-gap solution until rest of hh.org HAL stuff is merged.
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*/
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#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
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#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
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#ifdef CONFIG_SA1100_H3XXX
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#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
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#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
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#else
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#error This driver could serve H3x00 handhelds only!
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#endif
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static void sa11xx_uda1341_set_audio_clock(long val)
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{
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switch (val) {
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case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
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GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
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break;
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case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
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GPSR = GPIO_H3600_CLK_SET0;
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GPCR = GPIO_H3600_CLK_SET1;
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break;
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case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
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GPCR = GPIO_H3600_CLK_SET0;
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GPSR = GPIO_H3600_CLK_SET1;
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break;
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case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
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GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
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break;
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}
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}
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static void sa11xx_uda1341_set_samplerate(sa11xx_uda1341_t *sa11xx_uda1341, long rate)
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{
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int clk_div = 0;
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int clk=0;
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/* We don't want to mess with clocks when frames are in flight */
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Ser4SSCR0 &= ~SSCR0_SSE;
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/* wait for any frame to complete */
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udelay(125);
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/*
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* We have the following clock sources:
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* 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
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* Those can be divided either by 256, 384 or 512.
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* This makes up 12 combinations for the following samplerates...
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*/
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if (rate >= 48000)
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rate = 48000;
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else if (rate >= 44100)
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rate = 44100;
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else if (rate >= 32000)
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rate = 32000;
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else if (rate >= 29400)
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rate = 29400;
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else if (rate >= 24000)
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rate = 24000;
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else if (rate >= 22050)
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rate = 22050;
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else if (rate >= 21970)
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rate = 21970;
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else if (rate >= 16000)
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rate = 16000;
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else if (rate >= 14647)
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rate = 14647;
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else if (rate >= 10985)
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rate = 10985;
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else if (rate >= 10666)
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rate = 10666;
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else
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rate = 8000;
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/* Set the external clock generator */
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#ifdef CONFIG_H3600_HAL
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h3600_audio_clock(rate);
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#else
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sa11xx_uda1341_set_audio_clock(rate);
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#endif
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/* Select the clock divisor */
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switch (rate) {
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case 8000:
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case 10985:
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case 22050:
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case 24000:
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clk = F512;
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clk_div = SSCR0_SerClkDiv(16);
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break;
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case 16000:
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case 21970:
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case 44100:
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case 48000:
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clk = F256;
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clk_div = SSCR0_SerClkDiv(8);
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break;
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case 10666:
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case 14647:
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case 29400:
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case 32000:
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clk = F384;
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clk_div = SSCR0_SerClkDiv(12);
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break;
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}
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/* FMT setting should be moved away when other FMTs are added (FIXME) */
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l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
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l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
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Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
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sa11xx_uda1341->samplerate = rate;
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}
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/* }}} */
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/* {{{ HW init and shutdown */
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static void sa11xx_uda1341_audio_init(sa11xx_uda1341_t *sa11xx_uda1341)
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{
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unsigned long flags;
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/* Setup DMA stuff */
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sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
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sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
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sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
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sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
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sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
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sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
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/* Initialize the UDA1341 internal state */
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/* Setup the uarts */
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local_irq_save(flags);
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GAFR |= (GPIO_SSP_CLK);
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GPDR &= ~(GPIO_SSP_CLK);
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Ser4SSCR0 = 0;
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Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
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Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
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Ser4SSCR0 |= SSCR0_SSE;
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local_irq_restore(flags);
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/* Enable the audio power */
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#ifdef CONFIG_H3600_HAL
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h3600_audio_power(AUDIO_RATE_DEFAULT);
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#else
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clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
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set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
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set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
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#endif
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/* Wait for the UDA1341 to wake up */
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mdelay(1); //FIXME - was removed by Perex - Why?
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/* Initialize the UDA1341 internal state */
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l3_open(sa11xx_uda1341->uda1341);
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/* external clock configuration (after l3_open - regs must be initialized */
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sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
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/* Wait for the UDA1341 to wake up */
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set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
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mdelay(1);
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/* make the left and right channels unswapped (flip the WS latch) */
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Ser4SSDR = 0;
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#ifdef CONFIG_H3600_HAL
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h3600_audio_mute(0);
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#else
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clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
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#endif
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}
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static void sa11xx_uda1341_audio_shutdown(sa11xx_uda1341_t *sa11xx_uda1341)
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{
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/* mute on */
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#ifdef CONFIG_H3600_HAL
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h3600_audio_mute(1);
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#else
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set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
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#endif
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/* disable the audio power and all signals leading to the audio chip */
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l3_close(sa11xx_uda1341->uda1341);
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Ser4SSCR0 = 0;
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clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
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/* power off and mute off */
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/* FIXME - is muting off necesary??? */
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#ifdef CONFIG_H3600_HAL
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h3600_audio_power(0);
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h3600_audio_mute(0);
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#else
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clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
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clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
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#endif
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}
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/* }}} */
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/* {{{ DMA staff */
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/*
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* these are the address and sizes used to fill the xmit buffer
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* so we can get a clock in record only mode
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*/
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#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
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#define FORCE_CLOCK_SIZE 4096 // was 2048
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// FIXME Why this value exactly - wrote comment
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#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
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#ifdef HH_VERSION
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static int audio_dma_request(audio_stream_t *s, void (*callback)(void *, int))
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{
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int ret;
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ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
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if (ret < 0) {
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printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
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return ret;
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}
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sa1100_dma_set_callback(s->dmach, callback);
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return 0;
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}
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static inline void audio_dma_free(audio_stream_t *s)
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{
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sa1100_free_dma(s->dmach);
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s->dmach = -1;
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}
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#else
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static int audio_dma_request(audio_stream_t *s, void (*callback)(void *))
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{
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int ret;
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ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
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if (ret < 0)
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printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
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return ret;
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}
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static void audio_dma_free(audio_stream_t *s)
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{
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sa1100_free_dma((s)->dma_regs);
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(s)->dma_regs = 0;
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}
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#endif
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static u_int audio_get_dma_pos(audio_stream_t *s)
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{
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snd_pcm_substream_t * substream = s->stream;
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snd_pcm_runtime_t *runtime = substream->runtime;
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unsigned int offset;
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unsigned long flags;
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dma_addr_t addr;
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// this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
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spin_lock_irqsave(&s->dma_lock, flags);
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#ifdef HH_VERSION
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sa1100_dma_get_current(s->dmach, NULL, &addr);
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#else
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addr = sa1100_get_dma_pos((s)->dma_regs);
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#endif
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offset = addr - runtime->dma_addr;
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spin_unlock_irqrestore(&s->dma_lock, flags);
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offset = bytes_to_frames(runtime,offset);
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if (offset >= runtime->buffer_size)
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offset = 0;
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return offset;
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}
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|
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/*
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* this stops the dma and clears the dma ptrs
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*/
|
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static void audio_stop_dma(audio_stream_t *s)
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{
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unsigned long flags;
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spin_lock_irqsave(&s->dma_lock, flags);
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s->active = 0;
|
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s->period = 0;
|
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/* this stops the dma channel and clears the buffer ptrs */
|
|
#ifdef HH_VERSION
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sa1100_dma_flush_all(s->dmach);
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#else
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sa1100_clear_dma(s->dma_regs);
|
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#endif
|
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spin_unlock_irqrestore(&s->dma_lock, flags);
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}
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|
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static void audio_process_dma(audio_stream_t *s)
|
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{
|
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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) {
|
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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 = kzalloc(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_set_generic_dev(card)) < 0)
|
|
goto nodev;
|
|
|
|
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:
|
|
*/
|
|
|