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[/] [or1k_soc_on_altera_embedded_dev_kit/] [tags/] [linux-2.6/] [linux-2.6.24_or32_unified_v2.3/] [sound/] [arm/] [sa11xx-uda1341.c] - Rev 3
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/* * 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 <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> #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> #include <linux/l3/l3.h> #undef DEBUG_MODE #undef DEBUG_FUNCTION_NAMES #include <sound/uda1341.h> /* * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this * module for Familiar 0.6.1 */ /* {{{ Type definitions */ MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); static char *id; /* ID for this card */ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); struct audio_stream { char *id; /* identification string */ int stream_id; /* numeric identification */ dma_device_t dma_dev; /* device identifier for DMA */ #ifdef HH_VERSION dmach_t dmach; /* dma channel identification */ #else dma_regs_t *dma_regs; /* points to our DMA registers */ #endif unsigned int active:1; /* we are using this stream for transfer now */ int period; /* current transfer period */ int periods; /* current count of periods registerd in the DMA engine */ int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ unsigned int old_offset; spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ struct snd_pcm_substream *stream; }; struct sa11xx_uda1341 { struct snd_card *card; struct l3_client *uda1341; struct snd_pcm *pcm; long samplerate; struct audio_stream s[2]; /* playback & capture */ }; static unsigned int rates[] = { 8000, 10666, 10985, 14647, 16000, 21970, 22050, 24000, 29400, 32000, 44100, 48000, }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; static struct platform_device *device; /* }}} */ /* {{{ Clock and sample rate stuff */ /* * Stop-gap solution until rest of hh.org HAL stuff is merged. */ #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) #ifdef CONFIG_SA1100_H3XXX #define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) #define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) #else #error This driver could serve H3x00 handhelds only! #endif static void sa11xx_uda1341_set_audio_clock(long val) { switch (val) { case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; break; case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ GPSR = GPIO_H3600_CLK_SET0; GPCR = GPIO_H3600_CLK_SET1; break; case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ GPCR = GPIO_H3600_CLK_SET0; GPSR = GPIO_H3600_CLK_SET1; break; case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; break; } } static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate) { int clk_div = 0; int clk=0; /* We don't want to mess with clocks when frames are in flight */ Ser4SSCR0 &= ~SSCR0_SSE; /* wait for any frame to complete */ udelay(125); /* * We have the following clock sources: * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz * Those can be divided either by 256, 384 or 512. * This makes up 12 combinations for the following samplerates... */ if (rate >= 48000) rate = 48000; else if (rate >= 44100) rate = 44100; else if (rate >= 32000) rate = 32000; else if (rate >= 29400) rate = 29400; else if (rate >= 24000) rate = 24000; else if (rate >= 22050) rate = 22050; else if (rate >= 21970) rate = 21970; else if (rate >= 16000) rate = 16000; else if (rate >= 14647) rate = 14647; else if (rate >= 10985) rate = 10985; else if (rate >= 10666) rate = 10666; else rate = 8000; /* Set the external clock generator */ sa11xx_uda1341_set_audio_clock(rate); /* Select the clock divisor */ switch (rate) { case 8000: case 10985: case 22050: case 24000: clk = F512; clk_div = SSCR0_SerClkDiv(16); break; case 16000: case 21970: case 44100: case 48000: clk = F256; clk_div = SSCR0_SerClkDiv(8); break; case 10666: case 14647: case 29400: case 32000: clk = F384; clk_div = SSCR0_SerClkDiv(12); break; } /* FMT setting should be moved away when other FMTs are added (FIXME) */ l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; sa11xx_uda1341->samplerate = rate; } /* }}} */ /* {{{ HW init and shutdown */ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) { unsigned long flags; /* Setup DMA stuff */ sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; /* Initialize the UDA1341 internal state */ /* Setup the uarts */ local_irq_save(flags); GAFR |= (GPIO_SSP_CLK); GPDR &= ~(GPIO_SSP_CLK); Ser4SSCR0 = 0; Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; Ser4SSCR0 |= SSCR0_SSE; local_irq_restore(flags); /* Enable the audio power */ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); /* Wait for the UDA1341 to wake up */ mdelay(1); //FIXME - was removed by Perex - Why? /* Initialize the UDA1341 internal state */ l3_open(sa11xx_uda1341->uda1341); /* external clock configuration (after l3_open - regs must be initialized */ sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); /* Wait for the UDA1341 to wake up */ set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); mdelay(1); /* make the left and right channels unswapped (flip the WS latch) */ Ser4SSDR = 0; clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); } static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) { /* mute on */ set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); /* disable the audio power and all signals leading to the audio chip */ l3_close(sa11xx_uda1341->uda1341); Ser4SSCR0 = 0; clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); /* power off and mute off */ /* FIXME - is muting off necesary??? */ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); } /* }}} */ /* {{{ DMA staff */ /* * these are the address and sizes used to fill the xmit buffer * so we can get a clock in record only mode */ #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS #define FORCE_CLOCK_SIZE 4096 // was 2048 // FIXME Why this value exactly - wrote comment #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ #ifdef HH_VERSION static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int)) { int ret; ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); if (ret < 0) { printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); return ret; } sa1100_dma_set_callback(s->dmach, callback); return 0; } static inline void audio_dma_free(struct audio_stream *s) { sa1100_free_dma(s->dmach); s->dmach = -1; } #else static int audio_dma_request(struct audio_stream *s, void (*callback)(void *)) { int ret; ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); if (ret < 0) printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); return ret; } static void audio_dma_free(struct audio_stream *s) { sa1100_free_dma(s->dma_regs); s->dma_regs = 0; } #endif static u_int audio_get_dma_pos(struct audio_stream *s) { struct snd_pcm_substream *substream = s->stream; struct snd_pcm_runtime *runtime = substream->runtime; unsigned int offset; unsigned long flags; dma_addr_t addr; // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel spin_lock_irqsave(&s->dma_lock, flags); #ifdef HH_VERSION sa1100_dma_get_current(s->dmach, NULL, &addr); #else addr = sa1100_get_dma_pos((s)->dma_regs); #endif offset = addr - runtime->dma_addr; spin_unlock_irqrestore(&s->dma_lock, flags); offset = bytes_to_frames(runtime,offset); if (offset >= runtime->buffer_size) offset = 0; return offset; } /* * this stops the dma and clears the dma ptrs */ static void audio_stop_dma(struct audio_stream *s) { unsigned long flags; spin_lock_irqsave(&s->dma_lock, flags); s->active = 0; s->period = 0; /* this stops the dma channel and clears the buffer ptrs */ #ifdef HH_VERSION sa1100_dma_flush_all(s->dmach); #else sa1100_clear_dma(s->dma_regs); #endif spin_unlock_irqrestore(&s->dma_lock, flags); } static void audio_process_dma(struct audio_stream *s) { struct snd_pcm_substream *substream = s->stream; struct snd_pcm_runtime *runtime; unsigned int dma_size; unsigned int offset; int ret; /* we are requested to process synchronization DMA transfer */ if (s->tx_spin) { 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: */
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