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[alsa-devel] [PATCH] Coping with short replies in usbmixer
by thomas＠devbase.at 14 May '07

14 May '07

Hi, while trying to find out why a USB handset (Yealink P1K) frequently gets muted when changing volume, I found that short USB responses cause the problem. The function get_ctl_value(..) in usbmixer.c requests the state of a certain control and retries up to 10 times if an error occurs. However a short USB response is not treated as an error as neither the transfer flag URB_SHORT_NOT_OK is set nor is the return value of snd_usb_ctl_msg(..) compared with the requested amount of data. As a result the function get_ctl_value(..) might not retry the USB request but returns an unpredictable result without reporting an error. The attached patch forces the number of returned bytes to be at least val_len. It was generated from kernel version 2.6.21. If there are any issues with the patch please let me know. Otherwise I would be happy if it could be applied to the ALSA sources. Regards, -Thomas

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[alsa-devel] Meaning of this Plug PCM Transformation table?
by Nobin Mathew 14 May '07

14 May '07

Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 3840 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'IDENTITY' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 3840 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 This is capture Handle logs Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0.5 + 1*0.5 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'IDENTITY' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 I am having an application which does capture and playback simultaneously and is single threaded? What is the meaning of this Transformation table?

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[alsa-devel] [2/2] 2.6.22-rc1: known regressions
by Michal Piotrowski 13 May '07

13 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions (http://bugzilla.kernel.org/ is not available, information are not up-to-date) Networking Subject : panic with e1000 driver on HP Integrity servers References : http://bugzilla.kernel.org/show_bug.cgi?id=8455 Submitter : Doug Chapman <doug.chapman(a)hp.com> Caused-By : Auke Kok <auke-jan.h.kok(a)intel.com> commit e0aac5a289b1dacbc94bd9ae8c449bcdf9ab508c Status : problem is being debugged Timers/NOHZ Subject : 2.6.21-git4 BUG: soft lockup detected on CPU#1! References : http://lkml.org/lkml/2007/5/2/511 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Thomas Gleixner <tglx(a)linutronix.de> Status : problem is being debugged ALSA Subject : lost snd_4236 device References : http://lkml.org/lkml/2007/5/13/87 Submitter : Pete Clements <clem(a)clem.clem-digital.net> Status : Unknown Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] e-mu 1212m and 1820(m) patch
by Ctirad Fertr 13 May '07

13 May '07

Hello, here is a little patch which enables ADAT and S/PDIF functionality. It adds possibility to sync with either S/PDIF or ADAT clock as well. 44.1/48kHz only, though it works pretty well for me. Regards, Ctirad

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[alsa-devel] embedded sound architecture question
by Joachim Förster 13 May '07

13 May '07

Hi ALSA devs, I'm going to write an ALSA driver for a not yet existing AC97 controller, which is going to be "written" (VHDL), too (at the same time). Platform/Board is a Xilinx ML403 with Virtex-4 FPGA, PowerPC 405 architecture, OPB/OCP bus, AC97 Codec LM4550. Before presenting my question, I have to say, that I'm a beginner with ALSA/Linux driver development. My question is: Does the architecture described below make sense/is reasonable with ALSA and Linux? The problem is, that there is no DMA controller and implementing an OPB master device which would be able to do DMA itself is not an option at this time. So our thoughts were: Integrate the "DMA ring buffer" (which is usually somewhere in main memory/RAM) into the AC97 controller. Make ALSA access this HW buffer as if it was in main memory. This way, the device (AC97 controller) has "direct access" to its buffer "memory" - this could be called "fake DMA". The AC97 controller would have tell us where it is while playing and firing interrupts after one period, so that we don't write to values which are in the current period, instead update the area where of the past played periods etc. ... The buffer should be a "ring buffer", right? Mapping this "IO memory" into kernel space should be possible with io_remap_page_range(), right? I would have to implement the mmap() callback in my driver, to setup the given VMA (with the above function), right? So, ALSA library/applications will be able to use MMAP mode, which is what we want to achieve? [We don't want copy()/silence(). An intermediate buffer with ack()/tasklet/workqueue + FIFO in HW would be an alternative.] I read, that many applications don't work, if MMAP mode is not supported and classic read/write (copy()/silence()) is used, only. Is there a black list of apps, which don't work? Thanks for reading & your time, Joachim

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[alsa-devel] apologies for this email
by Adrian McMenamin 13 May '07

13 May '07

My last two made it in to the archive, but didn't get delivered to me. Did anyone else see them?

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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 12 May '07

12 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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