| /* |
| * 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 1.27 2005/12/07 09:13:42 cladisch 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/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 <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."); |
| |
| 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) */ |
| 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 */ |
| #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(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 */ |
| #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(struct sa11xx_uda1341 *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(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) { |
| 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 |
| { |
| 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 __init 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: |
| */ |