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[alsa-devel] Sound driver for Sega Dreamcast
by Adrian McMenamin 12 May '07

12 May '07

My last email is in the archive, but seem to make it back to me - in any case this code is better... but in case my last email didn't get delivered anywhere, I am asking for views on what needs to be done to get this committed. diff -Naur ./linux-2.6.21-orig/sound/Kconfig ./linux-2.6.21/sound/Kconfig --- ./linux-2.6.21-orig/sound/Kconfig 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Kconfig 2007-05-07 21:20:54.000000000 +0100 @@ -64,6 +64,8 @@ source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff -Naur ./linux-2.6.21-orig/sound/Makefile ./linux-2.6.21/sound/Makefile --- ./linux-2.6.21-orig/sound/Makefile 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Makefile 2007-05-07 21:37:05.000000000 +0100 @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff -Naur ./linux-2.6.21-orig/sound/sh/aica.c ./linux-2.6.21/sound/sh/aica.c --- ./linux-2.6.21-orig/sound/sh/aica.c 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.c 2007-05-12 19:10:32.000000000 +0100 @@ -0,0 +1,676 @@ +/* +* This code is licenced under +* the General Public Licence +* version 2 +* +* Copyright Adrian McMenamin 2005, 2006 +* <adrian(a)mcmen.demon.co.uk> +* See also http://newgolddream.dyndns.info/cgi-bin/cvsweb +* +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of version 2 of the GNU General Public License as published by +* the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#include <linux/init.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <linux/wait.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/timer.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <asm/io.h> +#include <asm/dma.h> +#include <asm/dreamcast/sysasic.h> +#include "aica.h" + +MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); +MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); + +/* module parameters */ +#define CARD_NAME "AICA" +static int index = -1; +static char *id; +static int enable = 1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param(enable, bool, 0644); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +/* Use workqueue */ + +static struct spu_work_holder { + struct work_struct spu_dma_work; + void *sspointer; +} spu_working; + +static struct workqueue_struct *aica_queue; + +/* Simple platform device */ +static struct platform_device *pd; +static struct resource aica_memory_space[2] = { + { + .name = "AICA ARM CONTROL", + .start = ARM_RESET_REGISTER, + .flags = IORESOURCE_MEM, + .end = ARM_RESET_REGISTER + 3, + }, + { + .name = "AICA Sound RAM", + .start = SPU_MEMORY_BASE, + .flags = IORESOURCE_MEM, + .end = SPU_MEMORY_BASE + 0x200000 - 1, + }, +}; + +/* SPU specific functions */ +/* spu_write_wait - wait for G2-SH FIFO to clear */ +static inline void spu_write_wait(void) +{ + int time_count; + time_count = 0; + while (1) { + if (!(readl(G2_FIFO) & 0x11)) + break; + /* To ensure hardware failure doesn't wedge kernel */ + time_count++; + if (time_count > 0x10000) + break; + } +} + +/* spu_memset - write to memory in SPU address space */ +static void spu_memset(uint32_t toi, uint32_t what, int length) +{ + int i; + snd_assert(length % 4 == 0, return); + spu_write_wait(); + for (i = 0; i < length; i++) { + writel(what, toi + SPU_MEMORY_BASE); + toi++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_memload - write to SPU address space */ +static void spu_memload(uint32_t toi, void __iomem * from, int length) +{ + uint32_t __iomem *froml = from; + uint32_t __iomem *to = (uint32_t __iomem *) (SPU_MEMORY_BASE + toi); + int i, val; + if (length % 4) + length = (length / 4) + 1; + else + length = length / 4; + spu_write_wait(); + for (i = 0; i < length; i++) { + val = *froml; + writel(val, to); + froml++; + to++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_disable - set spu registers to stop sound output */ +static void spu_disable(void) +{ + int i; + uint32_t regval; + spu_write_wait(); + regval = readl(ARM_RESET_REGISTER); + regval |= 1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); + for (i = 0; i < 64; i++) { + spu_write_wait(); + regval = readl(SPU_REGISTER_BASE + (i * 0x80)); + regval = (regval & ~0x4000) | 0x8000; + spu_write_wait(); + writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + } +} + +/* spu_enable - set spu registers to enable sound output */ +static void spu_enable(void) +{ + uint32_t regval = readl(ARM_RESET_REGISTER); + regval &= ~1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); +} + +/* + * Halt the sound processor, clear the memory, + * load some default ARM7 code, and then restart ARM7 +*/ +static void spu_reset(void) +{ + spu_disable(); + spu_memset(0, 0, 0x200000 / 4); + /* Put ARM7 in endless loop */ + ctrl_outl(0xea000002, SPU_MEMORY_BASE); + spu_enable(); +} + +/* aica_chn_start - write to spu to start playback */ +static void aica_chn_start(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_START, (uint32_t *) AICA_CONTROL_POINT); +} + +/* aica_chn_halt - write to spu to halt playback */ +static void aica_chn_halt(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_STOP, (uint32_t *) AICA_CONTROL_POINT); +} + +/* ALSA code below */ +static struct snd_pcm_hardware snd_pcm_aica_playback_hw = { + .info = (SNDRV_PCM_INFO_NONINTERLEAVED), + .formats = + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IMA_ADPCM), + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = AICA_BUFFER_SIZE, + .period_bytes_min = AICA_PERIOD_SIZE, + .period_bytes_max = AICA_PERIOD_SIZE, + .periods_min = AICA_PERIOD_NUMBER, + .periods_max = AICA_PERIOD_NUMBER, +}; + +static int aica_dma_transfer(int channels, int buffer_size, + struct snd_pcm_substream *substream) +{ + int q, err, period_offset; + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + err = 0; + dreamcastcard = substream->pcm->private_data; + period_offset = dreamcastcard->clicks; + period_offset %= (AICA_PERIOD_NUMBER / channels); + runtime = substream->runtime; + for (q = 0; q < channels; q++) { + err = dma_xfer(AICA_DMA_CHANNEL, + (unsigned long)(runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), + AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + + AICA_PERIOD_SIZE * period_offset, + buffer_size / channels, AICA_DMA_MODE); + if (unlikely(err < 0)) + break; + dma_wait_for_completion(AICA_DMA_CHANNEL); + } + return err; +} + +static void startup_aica(struct snd_card_aica *dreamcastcard) +{ + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + aica_chn_start(); +} + +static void run_spu_dma(struct work_struct *work) +{ + int buffer_size; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct snd_card_aica *dreamcastcard; + struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); + substream = holder-> sspointer; + dreamcastcard = substream->pcm->private_data; + runtime = substream->runtime; + if (unlikely(dreamcastcard->dma_check == 0)) { + buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels > 1) + dreamcastcard->channel->flags |= 0x01; + aica_dma_transfer(runtime->channels, buffer_size, substream); + startup_aica(dreamcastcard); + dreamcastcard->clicks = + buffer_size / (AICA_PERIOD_SIZE * runtime->channels); + return; + } else { + aica_dma_transfer(runtime->channels, + AICA_PERIOD_SIZE * runtime->channels, + substream); + snd_pcm_period_elapsed(dreamcastcard->substream); + dreamcastcard->clicks++; + dreamcastcard->clicks %= AICA_PERIOD_NUMBER; + mod_timer(&dreamcastcard->timer, jiffies + 1); + } +} + +static void aica_period_elapsed(unsigned long timer_var) +{ + /*timer fuction - so cannot sleep */ + int play_period; + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream; + struct snd_card_aica *dreamcastcard; + substream = (struct snd_pcm_substream *)timer_var; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Have we played out an additional period? */ + play_period = + frames_to_bytes(runtime, + readl + (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) / + AICA_PERIOD_SIZE; + if (play_period == dreamcastcard->current_period) { + /* reschedule the timer */ + mod_timer(&(dreamcastcard->timer), jiffies + 1); + return; + } + if (runtime->channels > 1) + dreamcastcard->current_period = play_period; + if (unlikely(dreamcastcard->dma_check == 0)) + dreamcastcard->dma_check = 1; + queue_work(aica_queue, &(spu_working.spu_dma_work)); +} + +static void spu_begin_dma(struct snd_pcm_substream *substream) +{ + /* Must be atomic */ + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Use queue to do the heavy lifting */ + spu_working.sspointer = substream; + INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); + queue_work(aica_queue, &(spu_working.spu_dma_work)); + /* Timer may already be running */ + if (unlikely(dreamcastcard->timer.data)) { + mod_timer(&dreamcastcard->timer, jiffies + 4); + return; + } + init_timer(&(dreamcastcard->timer)); + dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.function = aica_period_elapsed; + dreamcastcard->timer.expires = jiffies + 4; + add_timer(&(dreamcastcard->timer)); +} + +static int snd_aicapcm_pcm_open(struct snd_pcm_substream + *substream) +{ + struct snd_pcm_runtime *runtime; + struct aica_channel *channel; + struct snd_card_aica *dreamcastcard; + if (!enable) + return -ENOENT; + dreamcastcard = substream->pcm->private_data; + channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL); + if (!channel) + return -ENOMEM; + /* set defaults for channel */ + channel->sfmt = SM_8BIT; + channel->cmd = AICA_CMD_START; + channel->vol = dreamcastcard->master_volume; + channel->pan = 0x80; + channel->pos = 0; + channel->flags = 0; /* default to mono */ + dreamcastcard->channel = channel; + runtime = substream->runtime; + runtime->hw = snd_pcm_aica_playback_hw; + spu_enable(); + dreamcastcard->clicks = 0; + dreamcastcard->current_period = 0; + dreamcastcard->dma_check = 0; + return 0; +} + +static int snd_aicapcm_pcm_close(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + del_timer(&dreamcastcard->timer); + kfree(dreamcastcard->channel); + spu_disable(); + return 0; +} + +static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream + *substream) +{ + /* Free the DMA buffer */ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream + *substream, struct snd_pcm_hw_params + *hw_params) +{ + /* Allocate a DMA buffer using ALSA built-ins */ + return + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE) + dreamcastcard->channel->sfmt = SM_16BIT; + dreamcastcard->channel->freq = substream->runtime->rate; + dreamcastcard->substream = substream; + return 0; +} + +static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream + *substream, int cmd) +{ + struct snd_card_aica *dreamcastcard; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + spu_begin_dma(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + dreamcastcard = substream->pcm->private_data; + if (dreamcastcard->timer.data) + del_timer(&dreamcastcard->timer); + aica_chn_halt(); + break; + default: + return -EINVAL; + } + return 0; +} + +static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream + *substream) +{ + return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER); +} + +static struct snd_pcm_ops snd_aicapcm_playback_ops = { + .open = snd_aicapcm_pcm_open, + .close = snd_aicapcm_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aicapcm_pcm_hw_params, + .hw_free = snd_aicapcm_pcm_hw_free, + .prepare = snd_aicapcm_pcm_prepare, + .trigger = snd_aicapcm_pcm_trigger, + .pointer = snd_aicapcm_pcm_pointer, +}; + +/* TO DO: set up to handle more than one pcm instance */ +static int __init snd_aicapcmchip(struct snd_card_aica + *dreamcastcard, int pcm_index) +{ + struct snd_pcm *pcm; + int err; + /* AICA has no capture ability */ + err = + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + if (unlikely(err < 0)) + return err; + pcm->private_data = dreamcastcard; + strcpy(pcm->name, "AICA PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aicapcm_playback_ops); + /* Allocate the DMA buffers */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), + AICA_BUFFER_SIZE, + AICA_BUFFER_SIZE); + return err; +} + +/* Mixer controls */ +static int aica_pcmswitch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int aica_pcmswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */ + return 0; +} + +static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (ucontrol->value.integer.value[0] == 1) + return 0; /* TO DO: Fix me */ + else + aica_chn_halt(); + return 0; +} + +static int aica_pcmvolume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xFF; + return 0; +} + +static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; /* we've not yet been set up */ + ucontrol->value.integer.value[0] = dreamcastcard->channel->vol; + return 0; +} + +static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; + if (unlikely(dreamcastcard->channel->vol == + ucontrol->value.integer.value[0])) + return 0; + dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; + dreamcastcard->master_volume = ucontrol->value.integer.value[0]; + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + + return 1; +} + +static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .info = aica_pcmswitch_info, + .get = aica_pcmswitch_get, + .put = aica_pcmswitch_put +}; + +static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = aica_pcmvolume_info, + .get = aica_pcmvolume_get, + .put = aica_pcmvolume_put +}; + +static int load_aica_firmware(void) +{ + int err; + const struct firmware *fw_entry; + err = 0; + spu_reset(); + err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); + if (unlikely(err)) + return err; + /* write firware into memory */ + spu_disable(); + spu_memload(0, fw_entry->data, fw_entry->size); + spu_enable(); + release_firmware(fw_entry); + return err; +} + +static int __devinit add_aicamixer_controls(struct snd_card_aica + *dreamcastcard) +{ + int err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + return 0; +} + +static int snd_aica_remove(struct platform_device *devptr) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = platform_get_drvdata(devptr); + if (unlikely(!dreamcastcard)) + return -ENODEV; + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + platform_set_drvdata(devptr, NULL); + return 0; +} + +static int __init snd_aica_probe(struct platform_device *devptr) +{ + int err; + struct snd_card_aica *dreamcastcard; + dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + if (unlikely(!dreamcastcard)) + return -ENOMEM; + dreamcastcard->card = + snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); + if (unlikely(!dreamcastcard->card)) { + kfree(dreamcastcard); + return -ENODEV; + } + strcpy(dreamcastcard->card->driver, "snd_aica"); + strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); + strcpy(dreamcastcard->card->longname, + "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Load the PCM 'chip' */ + err = snd_aicapcmchip(dreamcastcard, 0); + if (unlikely(err < 0)) + goto freedreamcast; + snd_card_set_dev(dreamcastcard->card, &devptr->dev); + dreamcastcard->timer.data = 0; + dreamcastcard->channel = NULL; + /* Add basic controls */ + err = add_aicamixer_controls(dreamcastcard); + if (unlikely(err < 0)) + goto freedreamcast; + /* Register the card with ALSA subsystem */ + err = snd_card_register(dreamcastcard->card); + if (unlikely(err < 0)) + goto freedreamcast; + platform_set_drvdata(devptr, dreamcastcard); + aica_queue = create_workqueue("aica"); + if (unlikely(!aica_queue)) + goto freedreamcast; + snd_printk + ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + return 0; + freedreamcast: + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + return err; +} + +static struct platform_driver snd_aica_driver = { + .probe = snd_aica_probe, + .remove = snd_aica_remove, + .driver = { + .name = SND_AICA_DRIVER}, +}; + +static int __init aica_init(void) +{ + int err; + err = platform_driver_register(&snd_aica_driver); + if (unlikely(err < 0)) + return err; + pd = platform_device_register_simple(SND_AICA_DRIVER, -1, + aica_memory_space, 2); + if (unlikely(IS_ERR(pd))) { + platform_driver_unregister(&snd_aica_driver); + return PTR_ERR(pd); + } + /* Load the firmware */ + err = load_aica_firmware(); + if (unlikely(err < 0)) + return err; + + return 0; +} + +static void __exit aica_exit(void) +{ + /* Destroy the aica kernel thread */ + destroy_workqueue(aica_queue); + platform_device_unregister(pd); + platform_driver_unregister(&snd_aica_driver); + /* Kill any sound still playing and reset ARM7 to safe state */ + spu_reset(); +} + +module_init(aica_init); +module_exit(aica_exit); diff -Naur ./linux-2.6.21-orig/sound/sh/aica.h ./linux-2.6.21/sound/sh/aica.h --- ./linux-2.6.21-orig/sound/sh/aica.h 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.h 2007-05-07 21:47:31.000000000 +0100 @@ -0,0 +1,80 @@ +/* aica.h + * Header file for ALSA driver for + * Sega Dreamcast Yamaha AICA sound + * Copyright Adrian McMenamin + * <adrian(a)mcmen.demon.co.uk> + * 2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License as published by + * the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* SPU memory and register constants etc */ +#define G2_FIFO 0xa05f688c +#define SPU_MEMORY_BASE 0xA0800000 +#define ARM_RESET_REGISTER 0xA0702C00 +#define SPU_REGISTER_BASE 0xA0700000 + +/* AICA channels stuff */ +#define AICA_CONTROL_POINT 0xA0810000 +#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008 +#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004 + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +/* Buffer and period size */ +#define AICA_BUFFER_SIZE 0x8000 +#define AICA_PERIOD_SIZE 0x800 +#define AICA_PERIOD_NUMBER 16 + +#define AICA_CHANNEL0_OFFSET 0x11000 +#define AICA_CHANNEL1_OFFSET 0x21000 +#define CHANNEL_OFFSET 0x10000 + +#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_MODE 5 + +#define SND_AICA_DRIVER "AICA" + +struct aica_channel { + uint32_t cmd; /* Command ID */ + uint32_t pos; /* Sample position */ + uint32_t length; /* Sample length */ + uint32_t freq; /* Frequency */ + uint32_t vol; /* Volume 0-255 */ + uint32_t pan; /* Pan 0-255 */ + uint32_t sfmt; /* Sound format */ + uint32_t flags; /* Bit flags */ +}; + +struct snd_card_aica { + struct snd_card *card; + struct aica_channel *channel; + struct snd_pcm_substream *substream; + int clicks; + int current_period; + struct timer_list timer; + int master_volume; + int dma_check; +}; diff -Naur ./linux-2.6.21-orig/sound/sh/Kconfig ./linux-2.6.21/sound/sh/Kconfig --- ./linux-2.6.21-orig/sound/sh/Kconfig 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Kconfig 2007-05-07 21:28:30.000000000 +0100 @@ -0,0 +1,14 @@ +# ALSA SH drivers + +menu "SUPERH devices" + depends on SND!=n && SUPERH + +config SND_AICA + tristate "Dreamcast Yamaha AICA sound" + depends on SH_DREAMCAST && SND + select SND_PCM + help + ALSA Sound driver for the SEGA Dreamcast console. + +endmenu + diff -Naur ./linux-2.6.21-orig/sound/sh/Makefile ./linux-2.6.21/sound/sh/Makefile --- ./linux-2.6.21-orig/sound/sh/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Makefile 2007-05-07 21:34:10.000000000 +0100 @@ -0,0 +1,8 @@ +# +# Makefile for ALSA +# + +snd-aica-objs := aica.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_AICA) += snd-aica.o

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 11 May '07

11 May '07

o, I thought the epprom settings are default until the driver kicks in... So I set the jumpers accordingly and still no luck: IDESEL: 1-2 (Disabled - Its the CD IDE ON\OFF) JMPCFG: 1-2 (EPPROM ON) JMPB0: 1-2 (22XH) IDEIRQ: 1 (IRQ 15 - shouldn't make a different) JX1: 2-3 (Carbon MIC - again just letting you know though it shouldn't make a different) This should make sense with the picture I sent earlier... I also removed another card I had connected - ISA Modem, Just making sure their are no conflicts... Here's the grim results: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 FATAL: Error inserting snd_sb8 (/lib/modules/2.6.20-15-generic/kernel/sound/isa/sb/snd-sb8.ko): No such device P.S: Also an interesting note is that I now installed Win98 on top of that Dos and it worked too - It did also mention to use IRQ 11 so I also reserved in the bios, but again to no avail. Maybe I'm missing something ? Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] Mackie XD-2 Spike USB Drivers for Linux
by Sava Tatic 11 May '07

11 May '07

Hello, I have been trying to get the Mackie XD-2 Spike (http://www.mackie.com/products/spike/) to work on my Kubuntu Feisty machine, and so far I have only gotten it to play out (I had to use plughw instead of hw). I am using the generic USB drivers that came with the distro. I am having trouble recording anything on the card. Alsamixer (or Kmix) do not see the card. When I do alsamixer -c 1 (0 is my onboard card), it returns "no mixer elems found". I am aware that this card is not in the ALSA Matrix of supported cards, but I am wondering what would it take for it to become supported? All the best, Sava

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[alsa-devel] Capturing MIDI
by Steve Merrony 11 May '07

11 May '07

I hope this isn't too off-topic. I run an open source project (GENPO) which can be thought of as a kind of MIDI switchboard and filter. It uses ALSA throughout. I'd like to add a new feature to the software - a facility to simply record to a standard MIDI file anything that the program produces. I've searched though the API and I can't find anything documented which will help much, so here's my question... Is there a recommended library for C++ ALSA programs to use for handling (writing) standard MIDI files? Regards, Steve

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[alsa-devel] [PATCH2/5] ASoC S3C24xx machine drivers - Openmoko Neo1973
by Liam Girdwood 11 May '07

11 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds ALSA support for the Openmoko Neo1973 phone. Features:- * HiFi Playback and capture. * Phone calls supported. * Support for BT PCM in WM8753 voice interface. * Support for LM4857 audio amp. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH] ASoC AC97 device reg bugfix
by Liam Girdwood 11 May '07

11 May '07

This patch fixes a bug whereby AC97 bus device data was being clobbered when AC97 codecs using the generic ac97_codec.c driver were being registered. Codecs that didn't use the generic driver were unaffected (e.g. WM9712, WM9713). Changes:- o Add new AC97 codec class for custom (or need bus dev registration) AC97 codecs. o Only register/deregister this custom codec device with the AC97 bus. The generic AC97 driver already does this for generic codec devices. This may be related to bug #3038 :- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038 Manuel, can you try this on SH & Au1x00. Thanks Liam

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 10 May '07

10 May '07

Well, it didn't take long so here's the results : "sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1" gave no errors, and "lsmod" later gave: Module Size Used by snd_sb8 9216 0 snd_sb8_dsp 12288 1 snd_sb8 snd_sb_common 17792 2 snd_sb8,snd_sb8_dsp snd_opl3_lib 11520 1 snd_sb8 snd_hwdep 9988 1 snd_opl3_lib snd_pcm_oss 44544 0 snd_pcm 79876 3 snd_sb8,snd_sb8_dsp,snd_pcm_oss snd_mixer_oss 17408 1 snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 32896 0 snd_seq_midi 9600 0 snd_rawmidi 25472 2 snd_sb8_dsp,snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 52592 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 23684 3 snd_opl3_lib,snd_pcm,snd_seq snd_seq_device 9100 6 snd_opl3_lib,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54020 13 snd_sb8,snd_sb8_dsp,snd_sb_common,snd_opl3_lib,snd_hwdep,snd_pcm_oss,snd_pcm,snd_mixer_oss,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8672 1 snd snd_page_alloc 10888 1 snd_pcm nfs 240876 0 nfsd 218992 17 exportfs 6912 1 nfsd lockd 64904 3 nfs,nfsd sunrpc 161340 12 nfs,nfsd,lockd apm 22752 1 ppdev 10116 0 cpufreq_stats 7360 0 cpufreq_ondemand 9228 0 cpufreq_powersave 2688 0 freq_table 5792 2 cpufreq_stats,cpufreq_ondemand cpufreq_conservative 8200 0 cpufreq_userspace 5408 0 nls_iso8859_1 5120 1 nls_cp437 6784 1 vfat 14208 1 fat 53916 1 vfat ext2 66824 1 lp 12452 0 parport_pc 36388 1 parport 36936 3 ppdev,lp,parport_pc pcspkr 4224 0 ipv6 268704 8 tsdev 8768 0 evdev 11008 1 ext3 133128 1 jbd 59816 1 ext3 mbcache 9604 2 ext2,ext3 8139too 27648 0 ide_cd 32672 0 cdrom 37664 1 ide_cd ide_disk 17024 5 floppy 59524 0 ehci_hcd 34188 0 8139cp 25088 0 mii 6528 2 8139too,8139cp uhci_hcd 25360 0 usbcore 134280 3 ehci_hcd,uhci_hcd piix 10756 0 [permanent] generic 5124 0 [permanent] ata_generic 9092 0 libata 125720 1 ata_generic scsi_mod 142348 1 libata fbcon 42656 0 tileblit 3584 1 fbcon font 9216 1 fbcon bitblit 6912 1 fbcon softcursor 3200 1 bitblit vesafb 9220 0 capability 5896 0 commoncap 8192 1 capability But when I try to play an ogg gxine crushes an xmms ignores, when trying an mp3 gxine again crushes and xmms gives the soundcard no configured error... Just a recap: I verifyed using dos the settings (hwset and volset - default and save), and used the bios to set "IRQ 5" as "ISA Legacy". I didn't use the jumper as I see no need since the epprom already does the job. So I'm stuck, What do you think I should do now ? Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] detect extern/internel mic
by xinyun liu 10 May '07

10 May '07

It's interesting that when I plug in a headphone or an external microphone, the build in headphone and microphone will not work. Is it a driver feature or the hardware do it? Can anyone give some clue? I'd like to see the code but don't know how to start. Please give me help. Thx!

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Re: [alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 10 May '07

10 May '07

I have test all modes for alc883. The result is here: 0. 3stack-dig. both extern/internal can play, but can't record sound 1. 3stack-6ch-dig same with 3stack-dig 2. 3stack-6ch can play, but can't record sound. Can hear noise from the Mic. 3. 6stack-dig both can play sound, can't record sound 4. tarta-dig With alsamixer, can set to 2ch or 6ch, both can play sound, can't 5. tarta-2ch-dig both can play sound, can't record sound. 6. 6stack-dig-demo both can play sound, can't record sound. 7. aser both can play sound, can't record sound 8. medion can't play sound 9. laptop-eapd can play sound, but can't record sound 10. lenovo-101e both can play sound, can't record sound 11. auto both can play sound, and can record sound. But the sound quality is too bad to use from the internel Mic. After test it a while on switching the extern/internal Mic, the quality both droped. So I use lenovo-101e model, ( rebuild the driver, rmmod/insmod, use alsamixer to adjust the limits). Then I use auto mode, and the sound record work again from external Mic. After I switch to internel Mic, the quality droped again. Do you have any idea about it? Thx! 2007/4/30, Ustun ERGENOGLU <ustun.ergenoglu(a)gmail.com>: > I have a Fujitsu Siemens Amilo pa1510, with the same sound hardware, > and the same problem. I have posted some messages here, made a bug > report but no answer came :( i'm going hopeless. > > 2007/4/29, xinyun liu <xinyunliu.osc(a)gmail.com>: > > I know it's the system bios does not set the pin configuration right. > > But I don't know how to try out a correct model. > > > > Can anyone help to give some hints to fix it? > > > > > > Regards, > > Xinyun > > > > > > 2007/4/29, xinyun liu <xinyunliu.osc(a)gmail.com>: > > > hi, > > > I have an Acer Aspire 5100. I can play sound but with the build in > > > mic, I can't record sound. The driver is the latest HG code. > > > http://pastebin.ca/464406 > > > > > > [root@localhost ~]# lspci -nnv -d1002:437b > > > 00:14.2 0403: 1002:437b (rev 01) > > > Subsystem: 1025:009f > > > Flags: bus master, slow devsel, latency 64, IRQ 19 > > > Memory at c0000000 (64-bit, non-prefetchable) [size=16K] > > > Capabilities: [50] Power Management version 2 > > > Capabilities: [60] Message Signalled Interrupts: 64bit+ > > > Queue=0/0 Enable- > > > > > > > > > [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 > > > [root@localhost ~]# vi /tmp/1 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > > > codec_mask = 0xb > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' > > > is selected for config 1025:0 (Acer laptop) > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > > > codec_mask = 0xb > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' > > > is selected for config 1025:0 (Acer laptop) > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x8, stream=0x1, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x8, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x0, channel=0, format=0x0 > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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[alsa-devel] alsa-driver-1.0.14rc3 won't compile with kernel 2.6.20.3 when Processor family is Core2
by Chris Pemberton 10 May '07

10 May '07

Laptop has Pentium Dual Core and Intel HD Audio Kernel is 2.6.20.3 compiled for Processor family Core2. In-kernel alsa drivers are 1.0.14rc3 (cat /proc/asound/version) and compile ok. In-kernel drivers do not work for this sound card. Downloaded 1.0.14rc3 and applied patch_realtek.c ./configure --prefix=/usr --with-cards=hda-intel "make" gives the following error: make[1]: Entering directory `/usr/src/linux-2.6.20.3' > /usr/src/linux-2.6.20.3/Makefile:491: > /usr/src/linux-2.6.20.3/arch//Makefile: No such file or directory If the kernel is compiled with Processor family = Pentium M, the external alsa drivers compile clean and the soundcard works. So, Processor family = Core2 cannot be selected when using external alsa-driver-1.0.14rc3. Chris Pemberton

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