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Re: [alsa-devel] [RFC] ALSA vs. dedicated char device for a USB Audio Class gadget driver
by Laurent Pinchart 17 May '09

17 May '09

Hi Hal, first of all, thanks for your answer. On Thursday 14 May 2009 20:18:07 Hal Murray wrote: > > I need an API to transfer audio data from userspace to kernelspace. I > > initially thought about ALSA, but it turns out some assumptions made > > by ALSA are not fulfilled by my system. One of the most serious > > problems is that the UAC gadget driver doesn't have any audio clock. > > The only hardware clock available is the USB device controller > > interrupts generated at the USB transfer rate, and those are much > > faster than the audio sample rate. This will cause buffer underruns > > that I need to handle. > > You don't need a clock. The data will come to you at the right rate. All > you need to do is pass it on when you have enough to fill up a buffer. The > buffer size is fixed. It's part of the spec for the device you are > emulating. I'm not emulating any device. The buffer size is up to me, and I actually have a fixed number of small buffers, but that shouldn't make a difference. > Assume that you get samples one at a time each time the source clock > ticked. The normal state of your system would be to have a buffer that is > partially filled. When a new sample fills up the buffer, you would move it > from the input side to the ready-for-USB queue and setup a new buffer for > future input samples. > > Soon the USB side will read the queued buffer, you free that buffer and you > are now back to the normal state of collecting input data. > > You are in trouble if that doesn't happen before the nest buffer is ready, > that is the ready-for-USB queue should normally be empty. If you can't > keep it empty it will eventually overflow. Short chunks of time where the > queue builds up might be OK. For debugging, you should probably count > them. > > Now consider the case where you get several samples at a time rather than > one. Logically, copy them over one at a time. That turns into a burst of > source clocks, but the average will work out right. Your ready-for-USB > queue should still be empty most of the time. > > If that doesn't make sense, I'll try again. Consider something like audio > over a network. There is no audio clock on the network. The receiver can > derive the source clock by watching the data stream. This is more or less what I'm trying to do, with the difference that I don't move samples from the ALSA ring buffer when they arrive but when the USB layer asks for more data. My trouble is that I don't get notified when new samples are written to the ALSA ring buffer, and I'm not sure to find out how many samples are present in the buffer. If I could get some kind of "sample written" notification, or better, compute the size of data present in the ring buffer, my problem would be (mostly) solved. From what I understand, such a notification isn't possible when the ALSA ring buffer is mmap'ed, as ALSA itself doesn't get notified. This makes the second approach (reading the number of available samples) better, except that some kind of "free-wheeling" mode makes it impossible under some circumstances. From what Takashi Iwai explained in a mail to Jon Smirl on alsa-devel ("appl_ptr and DMA overrun at end of stream"), "we have also mode when appl_ptr is not updated at all (when stop_threshold == boundary)". This free-wheeling mode seems to be used by dmix. Best regards, Laurent Pinchart

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[alsa-devel] [PATCH] ASoC: Add TXx9 AC link controller driver
by Atsushi Nemoto 17 May '09

17 May '09

This patch adds support for the integrated ACLC of the TXx9 family. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- This patch depends on txx9dmac patch (driver) in linux-mips queue tree or linux-next tree. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/txx9/Kconfig | 29 +++ sound/soc/txx9/Makefile | 11 + sound/soc/txx9/txx9aclc-ac97.c | 214 +++++++++++++++++++ sound/soc/txx9/txx9aclc-generic.c | 129 ++++++++++++ sound/soc/txx9/txx9aclc.c | 415 +++++++++++++++++++++++++++++++++++++ sound/soc/txx9/txx9aclc.h | 74 +++++++ 8 files changed, 874 insertions(+), 0 deletions(-) create mode 100644 sound/soc/txx9/Kconfig create mode 100644 sound/soc/txx9/Makefile create mode 100644 sound/soc/txx9/txx9aclc-ac97.c create mode 100644 sound/soc/txx9/txx9aclc-generic.c create mode 100644 sound/soc/txx9/txx9aclc.c create mode 100644 sound/soc/txx9/txx9aclc.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3d2bb6f..85e985a 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/txx9/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..6039b0d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,3 +11,4 @@ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig new file mode 100644 index 0000000..ebc9327 --- /dev/null +++ b/sound/soc/txx9/Kconfig @@ -0,0 +1,29 @@ +## +## TXx9 ACLC +## +config SND_SOC_TXX9ACLC + tristate "SoC Audio for TXx9" + depends on HAS_TXX9_ACLC && TXX9_DMAC + help + This option enables support for the AC Link Controllers in TXx9 SoC. + +config HAS_TXX9_ACLC + bool + +config SND_SOC_TXX9ACLC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_TXX9ACLC_GENERIC + tristate "Generic TXx9 ACLC sound machine" + depends on SND_SOC_TXX9ACLC + select SND_SOC_TXX9ACLC_AC97 + select SND_SOC_AC97_CODEC + help + This is a generic AC97 sound machine for use in TXx9 based systems. diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile new file mode 100644 index 0000000..551f16c --- /dev/null +++ b/sound/soc/txx9/Makefile @@ -0,0 +1,11 @@ +# Platform +snd-soc-txx9aclc-objs := txx9aclc.o +snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o +obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o + +# Machine +snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c new file mode 100644 index 0000000..5bf9fd0 --- /dev/null +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -0,0 +1,214 @@ +/* + * TXx9 ACLC AC97 driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include "txx9aclc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#ifdef __BIG_ENDIAN +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_BE +#else +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_LE +#endif + +static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq); + +/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */ +static struct txx9aclc_soc_device *txx9aclc_soc_dev; + +static int txx9aclc_regready(struct txx9aclc_soc_device *dev) +{ + return __raw_readl(dev->base + ACINTSTS) & ACINT_REGACCRDY; +} + +/* AC97 controller reads codec register */ +static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + u32 dat; + + if (!(__raw_readl(dev->base + ACINTSTS) & ACINT_CODECRDY(ac97->num))) + return 0xffff; + reg |= ac97->num << 7; + dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ; + __raw_writel(dat, dev->base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, dev->base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + __raw_writel(ACINT_REGACCRDY, dev->base + ACINTDIS); + dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg); + dat = 0xffff; + goto done; + } + dat = __raw_readl(dev->base + ACREGACC); + if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) { + dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n", + dat, reg); + dat = 0xffff; + goto done; + } + dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff; +done: + __raw_writel(ACINT_REGACCRDY, dev->base + ACINTDIS); + return dat; +} + +/* AC97 controller writes to codec register */ +static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + + __raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) | + (val << ACREGACC_DAT_SHIFT), + dev->base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, dev->base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + dev_err(dev->soc_dev.dev, + "ac97 write timeout (reg %#x)\n", reg); + } + __raw_writel(ACINT_REGACCRDY, dev->base + ACINTDIS); +} + +static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY; + + __raw_writel(ACCTL_ENLINK, dev->base + ACCTLDIS); + mmiowb(); + udelay(1); + __raw_writel(ACCTL_ENLINK, dev->base + ACCTLEN); + /* wait for primary codec ready status */ + __raw_writel(ready, dev->base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, + (__raw_readl(dev->base + ACINTSTS) & ready) + == ready, HZ)) { + dev_err(&ac97->dev, "primary codec is not ready " + "(status %#x)\n", + __raw_readl(dev->base + ACINTSTS)); + } + __raw_writel(ACINT_REGACCRDY, dev->base + ACINTSTS); + __raw_writel(ready, dev->base + ACINTDIS); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = txx9aclc_ac97_read, + .write = txx9aclc_ac97_write, + .reset = txx9aclc_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) +{ + struct txx9aclc_soc_device *dev = dev_id; + + __raw_writel(__raw_readl(dev->base + ACINTMSTS), dev->base + ACINTDIS); + wake_up(&ac97_waitq); + return IRQ_HANDLED; +} + +static int txx9aclc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + int err; + + err = request_irq(dev->irq, txx9aclc_ac97_irq, + IRQF_DISABLED, dev_name(&pdev->dev), dev); + if (err < 0) + return err; + txx9aclc_soc_dev = dev; + return 0; +} + +static void txx9aclc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + + /* disable AC-link */ + __raw_writel(ACCTL_ENLINK, dev->base + ACCTLDIS); + free_irq(dev->irq, dev); + txx9aclc_soc_dev = NULL; +} + +#ifdef CONFIG_PM +static int txx9aclc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int txx9aclc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} +#else +#define txx9aclc_ac97_suspend NULL +#define txx9aclc_ac97_resume NULL +#endif + +struct snd_soc_dai txx9aclc_ac97_dai = { + .name = "txx9aclc_ac97", + .ac97_control = 1, + .probe = txx9aclc_ac97_probe, + .remove = txx9aclc_ac97_remove, + .suspend = txx9aclc_ac97_suspend, + .resume = txx9aclc_ac97_resume, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, +}; +EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai); + +static int __init txx9aclc_ac97_init(void) +{ + return snd_soc_register_dai(&txx9aclc_ac97_dai); +} + +static void __exit txx9aclc_ac97_exit(void) +{ + snd_soc_unregister_dai(&txx9aclc_ac97_dai); +} + +module_init(txx9aclc_ac97_init); +module_exit(txx9aclc_ac97_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c new file mode 100644 index 0000000..edd1c37 --- /dev/null +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -0,0 +1,129 @@ +/* + * Generic TXx9 ACLC machine driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at ACLC (e.g. RBTX49XX boards). + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include "../codecs/ac97.h" +#include "txx9aclc.h" + +static struct snd_soc_dai_link txx9aclc_generic_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &txx9aclc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card txx9aclc_generic_card = { + .name = "Generic TXx9 ACLC Audio", + .platform = &txx9aclc_soc_platform, + .dai_link = &txx9aclc_generic_dai, + .num_links = 1, +}; + +static struct txx9aclc_soc_device txx9aclc_generic_soc_device = { + .soc_dev = { + .card = &txx9aclc_generic_card, + .codec_dev = &soc_codec_dev_ac97, + }, +}; + +static int __init txx9aclc_generic_probe(struct platform_device *pdev) +{ + struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device; + struct platform_device *soc_pdev; + struct resource *r; + int ret; + int i; + int irq; + + irq = platform_get_irq(pdev, 0); + if (irq < 0) + return irq; + dev->irq = irq; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) + return -EBUSY; + + if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), + dev_name(&pdev->dev))) + return -EBUSY; + dev->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); + if (!dev->base) + return -EBUSY; + dev->physbase = r->start; + if (sizeof(dev->physbase) > sizeof(r->start) && + r->start >= TXX9_DIRECTMAP_BASE && + r->start < TXX9_DIRECTMAP_BASE + 0x400000) + dev->physbase |= 0xf00000000ull; + + for (i = 0; i < 2; i++) { + r = platform_get_resource(pdev, IORESOURCE_DMA, i); + if (!r) + return -EBUSY; + dev->dmadata[i].dma_res = r; + } + + soc_pdev = platform_device_alloc("soc-audio", -1); + if (!soc_pdev) + return -ENOMEM; + platform_set_drvdata(soc_pdev, &dev->soc_dev); + dev->soc_dev.dev = &soc_pdev->dev; + ret = platform_device_add(soc_pdev); + if (ret) { + platform_device_put(soc_pdev); + return ret; + } + platform_set_drvdata(pdev, soc_pdev); + return 0; +} + +static int __exit txx9aclc_generic_remove(struct platform_device *pdev) +{ + struct platform_device *soc_pdev = platform_get_drvdata(pdev); + + platform_device_unregister(soc_pdev); + return 0; +} + +static struct platform_driver txx9aclc_generic_driver = { + .remove = txx9aclc_generic_remove, + .driver = { + .name = "txx9aclc-generic", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_generic_init(void) +{ + return platform_driver_probe(&txx9aclc_generic_driver, + txx9aclc_generic_probe); +} + +static void __exit txx9aclc_generic_exit(void) +{ + platform_driver_unregister(&txx9aclc_generic_driver); +} + +module_init(txx9aclc_generic_init); +module_exit(txx9aclc_generic_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c new file mode 100644 index 0000000..0dc3153 --- /dev/null +++ b/sound/soc/txx9/txx9aclc.c @@ -0,0 +1,415 @@ +/* + * Generic TXx9 ACLC platform driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/scatterlist.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "txx9aclc.h" + +static const struct snd_pcm_hardware txx9aclc_pcm_hardware = { + /* + * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID + * needs more works for noncoherent MIPS. + */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE, +#ifdef __BIG_ENDIAN + .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif + .period_bytes_min = 1024, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 4096, + .buffer_bytes_max = 32 * 1024, +}; + +static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_device *socdev = rtd->socdev; + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + dev_dbg(socdev->dev, + "runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd " + "runtime->min_align %ld\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + dev_dbg(socdev->dev, + "periods %d period_bytes %d stream %d\n", + params_periods(params), params_period_bytes(params), + substream->stream); + + dmadata->substream = substream; + dmadata->pos = 0; + return 0; +} + +static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + + dmadata->dma_addr = runtime->dma_addr; + dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + dmadata->period_bytes = snd_pcm_lib_period_bytes(substream); + + if (dmadata->buffer_bytes == dmadata->period_bytes) { + dmadata->frag_bytes = dmadata->period_bytes >> 1; + dmadata->frags = 2; + } else { + dmadata->frag_bytes = dmadata->period_bytes; + dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes; + } + dmadata->frag_count = 0; + dmadata->pos = 0; + return 0; +} + +static void txx9aclc_dma_complete(void *arg) +{ + struct txx9aclc_dmadata *dmadata = arg; + unsigned long flags; + + /* dma completion handler cannot submit new operations */ + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count >= 0) { + dmadata->dmacount--; + BUG_ON(dmadata->dmacount < 0); + tasklet_schedule(&dmadata->tasklet); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static struct dma_async_tx_descriptor * +txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) +{ + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct scatterlist sg; + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)), + dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1)); + sg_dma_address(&sg) = buf_dma_addr; + desc = chan->device->device_prep_slave_sg(chan, &sg, 1, + dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return NULL; + } + desc->callback = txx9aclc_dma_complete; + desc->callback_param = dmadata; + desc->tx_submit(desc); + return desc; +} + +#define NR_DMA_CHAIN 2 + +static void txx9aclc_dma_tasklet(unsigned long data) +{ + struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data; + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct snd_pcm_substream *substream = dmadata->substream; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + int i; + unsigned long flags; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count < 0) { + struct txx9aclc_soc_device *dev = + container_of(dmadata, struct txx9aclc_soc_device, + dmadata[substream->stream]); + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + chan->device->device_terminate_all(chan); + /* first time */ + for (i = 0; i < NR_DMA_CHAIN; i++) { + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + i * dmadata->frag_bytes); + if (!desc) + return; + } + dmadata->dmacount = NR_DMA_CHAIN; + chan->device->device_issue_pending(chan); + spin_lock_irqsave(&dmadata->dma_lock, flags); + __raw_writel(ctlbit, dev->base + ACCTLEN); + dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return; + } + BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN); + while (dmadata->dmacount < NR_DMA_CHAIN) { + dmadata->dmacount++; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + + dmadata->frag_count * dmadata->frag_bytes); + if (!desc) + return; + chan->device->device_issue_pending(chan); + + spin_lock_irqsave(&dmadata->dma_lock, flags); + dmadata->frag_count++; + dmadata->frag_count %= dmadata->frags; + dmadata->pos += dmadata->frag_bytes; + dmadata->pos %= dmadata->buffer_bytes; + if ((dmadata->frag_count * dmadata->frag_bytes) % + dmadata->period_bytes == 0) + snd_pcm_period_elapsed(substream); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + unsigned long flags; + int ret = 0; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dmadata->frag_count = -1; + tasklet_schedule(&dmadata->tasklet); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + __raw_writel(ctlbit, dev->base + ACCTLDIS); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + __raw_writel(ctlbit, dev->base + ACCTLEN); + break; + default: + ret = -EINVAL; + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return ret; +} + +static snd_pcm_uframes_t +txx9aclc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + + return bytes_to_frames(substream->runtime, dmadata->pos); +} + +static int txx9aclc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream]; + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware); + if (ret) + return ret; + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + substream->runtime->private_data = dmadata; + return 0; +} + +static int txx9aclc_pcm_close(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct dma_chan *chan = dmadata->dma_chan; + + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + return 0; +} + +static struct snd_pcm_ops txx9aclc_pcm_ops = { + .open = txx9aclc_pcm_open, + .close = txx9aclc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = txx9aclc_pcm_hw_params, + .hw_free = txx9aclc_pcm_hw_free, + .prepare = txx9aclc_pcm_prepare, + .trigger = txx9aclc_pcm_trigger, + .pointer = txx9aclc_pcm_pointer, +}; + +static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, 64 * 1024, 4 * 1024 * 1024); +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct txx9aclc_dmadata *dmadata = param; + char devname[BUS_ID_SIZE + 2]; + + sprintf(devname, "%s.%d", dmadata->dma_res->name, + (int)dmadata->dma_res->start); + if (strcmp(dev_name(chan->device->dev), devname) == 0) { + chan->private = &dmadata->dma_slave; + return true; + } + return false; +} + +static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, + struct txx9aclc_dmadata *dmadata) +{ + struct txx9dmac_slave *ds = &dmadata->dma_slave; + dma_cap_mask_t mask; + + spin_lock_init(&dmadata->dma_lock); + + ds->reg_width = sizeof(u32); + if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ds->tx_reg = dev->physbase + ACAUDODAT; + ds->rx_reg = 0; + } else { + ds->tx_reg = 0; + ds->rx_reg = dev->physbase + ACAUDIDAT; + } + + /* Try to grab a DMA channel */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dmadata->dma_chan = dma_request_channel(mask, filter, dmadata); + if (!dmadata->dma_chan) { + dev_err(dev->soc_dev.dev, + "DMA channel for %s is not available\n", + dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return -EBUSY; + } + tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet, + (unsigned long)dmadata); + return 0; +} + +static int txx9aclc_pcm_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + int i; + int ret; + + dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK; + dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE; + for (i = 0; i < 2; i++) { + ret = txx9aclc_dma_init(dev, &dev->dmadata[i]); + if (ret) + goto exit; + } + return 0; + +exit: + for (i = 0; i < 2; i++) { + if (dev->dmadata[i].dma_chan) + dma_release_channel(dev->dmadata[i].dma_chan); + dev->dmadata[i].dma_chan = NULL; + } + return ret; +} + +static int txx9aclc_pcm_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + int i; + + /* disable all FIFO DMAs */ + __raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, dev->base + ACCTLDIS); + /* dummy R/W to clear pending DMAREQ if any */ + __raw_writel(__raw_readl(dev->base + ACAUDIDAT), + dev->base + ACAUDODAT); + + for (i = 0; i < 2; i++) { + struct txx9aclc_dmadata *dmadata = &dev->dmadata[i]; + struct dma_chan *chan = dmadata->dma_chan; + if (chan) { + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + dma_release_channel(chan); + } + dev->dmadata[i].dma_chan = NULL; + } + return 0; +} + +struct snd_soc_platform txx9aclc_soc_platform = { + .name = "txx9aclc-audio", + .probe = txx9aclc_pcm_probe, + .remove = txx9aclc_pcm_remove, + .pcm_ops = &txx9aclc_pcm_ops, + .pcm_new = txx9aclc_pcm_new, + .pcm_free = txx9aclc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(txx9aclc_soc_platform); + +static int __init txx9aclc_soc_platform_init(void) +{ + return snd_soc_register_platform(&txx9aclc_soc_platform); +} + +static void __exit txx9aclc_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&txx9aclc_soc_platform); +} + +module_init(txx9aclc_soc_platform_init); +module_exit(txx9aclc_soc_platform_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h new file mode 100644 index 0000000..c1c1162 --- /dev/null +++ b/sound/soc/txx9/txx9aclc.h @@ -0,0 +1,74 @@ +/* + * TXx9 SoC AC Link Controller + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __TXX9ACLC_H +#define __TXX9ACLC_H + +#include <linux/interrupt.h> +#include <asm/txx9/dmac.h> + +#define ACCTLEN 0x00 /* control enable */ +#define ACCTLDIS 0x04 /* control disable */ +#define ACCTL_ENLINK 0x00000001 /* enable/disable AC-link */ +#define ACCTL_AUDODMA 0x00000100 /* AUDODMA enable/disable */ +#define ACCTL_AUDIDMA 0x00001000 /* AUDIDMA enable/disable */ +#define ACCTL_AUDOEHLT 0x00010000 /* AUDO error halt + enable/disable */ +#define ACCTL_AUDIEHLT 0x00100000 /* AUDI error halt + enable/disable */ +#define ACREGACC 0x08 /* codec register access */ +#define ACREGACC_DAT_SHIFT 0 /* data field */ +#define ACREGACC_REG_SHIFT 16 /* address field */ +#define ACREGACC_CODECID_SHIFT 24 /* CODEC ID field */ +#define ACREGACC_READ 0x80000000 /* CODEC read */ +#define ACREGACC_WRITE 0x00000000 /* CODEC write */ +#define ACINTSTS 0x10 /* interrupt status */ +#define ACINTMSTS 0x14 /* interrupt masked status */ +#define ACINTEN 0x18 /* interrupt enable */ +#define ACINTDIS 0x1c /* interrupt disable */ +#define ACINT_CODECRDY(n) (0x00000001 << (n)) /* CODECn ready */ +#define ACINT_REGACCRDY 0x00000010 /* ACREGACC ready */ +#define ACINT_AUDOERR 0x00000100 /* AUDO underrun error */ +#define ACINT_AUDIERR 0x00001000 /* AUDI overrun error */ +#define ACDMASTS 0x80 /* DMA request status */ +#define ACDMA_AUDO 0x00000001 /* AUDODMA pending */ +#define ACDMA_AUDI 0x00000010 /* AUDIDMA pending */ +#define ACAUDODAT 0xa0 /* audio out data */ +#define ACAUDIDAT 0xb0 /* audio in data */ +#define ACREVID 0xfc /* revision ID */ + +struct txx9aclc_dmadata { + struct resource *dma_res; + struct txx9dmac_slave dma_slave; + struct dma_chan *dma_chan; + struct tasklet_struct tasklet; + spinlock_t dma_lock; + int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */ + struct snd_pcm_substream *substream; + unsigned long pos; + dma_addr_t dma_addr; + unsigned long buffer_bytes; + unsigned long period_bytes; + unsigned long frag_bytes; + int frags; + int frag_count; + int dmacount; +}; + +struct txx9aclc_soc_device { + struct snd_soc_device soc_dev; + void __iomem *base; + u64 physbase; + int irq; + struct txx9aclc_dmadata dmadata[2]; +}; + +extern struct snd_soc_platform txx9aclc_soc_platform; +extern struct snd_soc_dai txx9aclc_ac97_dai; + +#endif /* __TXX9ACLC_H */ -- 1.5.6.5

2 6

[alsa-devel] Another successful X-Fi driver test.
by A.Haq Abbad 16 May '09

16 May '09

Here are some details: System: Intel Core 2 Duo E6600 Asus P5W DH Motherboard X-Fi eXtreme Music Intel HD audio onboard Software: Advanced Linux Sound Architecture Driver Version 1.0.20. Compiled on May 16 2009 for kernel 2.6.28-11-generic (SMP). Running Kubuntu 9.04, with default packages. All major audio players tested and work fine. Aduiocious does not work with 24bit mode as reported before. System is stable, no issues encountered yet! Recording through the mic works flawlessly. Sound from speakers seem clear. Switching between intel HD onboard audio and xfi works fine, both enabled on the same system. P.S I installed the file alsa-driver-unstable-snapshot.tar.gz now my alsa version is 1.0.20 instead of .18, does that cause major issues? I thought the package will only install the module snd-ctxfi, not the whole alsa! or is that not possible. Let me know if you want me to run specific test. Regards.

2 1

[alsa-devel] SiS SI7012 with ALC202 sound card not working on Acer Aspire G600 in Ubuntu 9.04
by puli dilip 16 May '09

16 May '09

SiS SI7012 with ALC202 sound card not working on Acer Aspire G600 in Ubuntu 9.04 Card: SiS SI7012 │ │ Chip: Realtek ALC202 rev 0 this is taken from alsamixer 00:00.0 Host bridge: Silicon Integrated Systems [SiS] 651 Host (rev 01) 00:01.0 PCI bridge: Silicon Integrated Systems [SiS] Virtual PCI-to-PCI bridge (AGP) 00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS962 [MuTIOL Media IO] (rev 04) 00:02.1 SMBus: Silicon Integrated Systems [SiS] SiS961/2 SMBus Controller 00:02.3 FireWire (IEEE 1394): Silicon Integrated Systems [SiS] FireWire Controller 00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] AC'97 Sound Controller (rev a0) 00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.1 Controller (rev 0f) 00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.1 Controller (rev 0f) 00:03.2 USB Controller: Silicon Integrated Systems [SiS] USB 1.1 Controller (rev 0f) 00:03.3 USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller 00:06.0 Multimedia controller: Philips Semiconductors SAA7130 Video Broadcast Decoder (rev 01) 00:0f.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) 01:00.0 VGA compatible controller: Silicon Integrated Systems [SiS] 65x/M650/740 PCI/AGP VGA Display Adapter here i am giving link please fix the problem https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/70972 Explore and discover exciting holidays and getaways with Yahoo! India Travel http://in.travel.yahoo.com/

2 1

[alsa-devel] FIXED by substituting 1.0.19 pcm_lib.c for 1.0.21 pcm_lib.c -> Re: ALSA 1.0.20 PCM Capture Problem (Does not happen on ALSA 1.0.19)
by Robert Krakora 16 May '09

16 May '09

I grabbed ALSA 1.0.20 and attempted to capture video from my Creative USB webcam and audio from my analog audio microphone. To my surprise there was audio being captured but only in bits and pieces, nothing audible. If I recorded the audio and played it back it sound like broken audio. I noticed this in the release notes: [ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies I assumed it had something to do with changes made in pcm_lib.c, so I replaced the ALSA 1.0.20 version of this file with the ALSA 1.0.19 version, rebuilt and I am now able to capture audio again from my analog microphone. I filed Bug Report 0004523 describing this problem. On Tue, May 12, 2009 at 5:16 PM, Robert Vincent Krakora <rob.krakora(a)messagenetsystems.com> wrote: > Hello, > > I am utilizing an external mic on my machine in conjunction with a > VF0560 Live! Cam Optia AF webcam. I am running CentOS 5.3 with the > kernel listed below and ALSA 1.0.20. I using the machine to stream > video with VLC 9.9 (see further below). VLC spits out the following on > the console of my machine whist capturing A/V and serving it up on the > network...ALSA 1.0.19 did not exhibit this problem while streaming it > did lock up after 20 minutes or so (audio could be heard on the VLC or > mplayer client with ALSA 1.0.19...just bits and pieces of audio every > now and then with ALSA 1.0.20. V4L is not the culprit here as video > streams and can be received without incident if I do not also capture > audio. The mic is plugged into the onboard sound on the motherboard. I > have not yet tried my USB mic with ALSA 1.0.20, but I will try it later > tonight. Any and all help would be greatly appreciate. Thanks in advance. > > [00000473] main audio output warning: audio drift is too big (-146765), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29715) > [00000473] main audio output warning: audio drift is too big (-150692), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30980) > [00000493] main spu text debug: looking for text renderer module: 3 > candidates > [00000493] freetype spu text error: failed to load font file > /usr/share/fonts/truetype/freefont/FreeSerifBold.ttf > [00000493] main spu text debug: using text renderer module "svg" > [00000493] main spu text debug: TIMER module_Need() : 29.750 ms - Total > 29.750 ms / 1 intvls (Avg 29.750 ms) > [00000492] main blend debug: looking for video blending module: 1 candidate > [00000492] blend blend debug: chroma: YUVA -> I420 > [00000492] main blend debug: using video blending module "blend" > [00000492] main blend debug: TIMER module_Need() : 0.433 ms - Total > 0.433 ms / 1 intvls (Avg 0.433 ms) > [00000473] main audio output warning: audio drift is too big (-147017), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29763) > [00000473] main audio output warning: audio drift is too big (-147157), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30302) > [00000473] main audio output warning: audio drift is too big (-150434), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30931) > [00000473] main audio output warning: buffer is 114796 in advance, > triggering downsampling > [00000473] main audio output warning: audio drift is too big (-261628), > clearing out > [00000473] main audio output warning: timing screwed, stopping resampling > [00000473] main audio output warning: mixer start isn't output start > (-53788) > [00000473] main audio output warning: audio drift is too big (-146580), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29711) > [00000473] main audio output warning: audio drift is too big (-150686), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30979) > [00000473] main audio output warning: audio drift is too big (-146733), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29708) > [00000473] main audio output warning: audio drift is too big (-146978), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30267) > [00000473] main audio output warning: audio drift is too big (-150569), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30445) > [00000473] main audio output warning: buffer is 114650 in advance, > triggering downsampling > [00000473] main audio output warning: audio drift is too big (-261447), > clearing out > [00000473] main audio output warning: timing screwed, stopping resampling > [00000473] main audio output warning: mixer start isn't output start > (-54265) > [00000473] main audio output warning: audio drift is too big (-146518), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29699) > [00000473] main audio output warning: audio drift is too big (-146731), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30220) > [00000473] main audio output warning: audio drift is too big (-150880), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30505) > [00000473] main audio output warning: audio drift is too big (-146644), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30203) > [00000473] main audio output warning: audio drift is too big (-146701), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29702) > [00000473] main audio output warning: audio drift is too big (-150683), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30979) > [00000473] main audio output warning: audio drift is too big (-146706), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29703) > [00000473] main audio output warning: audio drift is too big (-146916), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30255) > [00000473] main audio output warning: audio drift is too big (-150819), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-31005) > [00000473] main audio output warning: audio drift is too big (-147003), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-29760) > [00000473] main audio output warning: audio drift is too big (-146886), > clearing out > [00000473] main audio output warning: mixer start isn't output start > (-30250) > [00000473] main audio output warning: audio drift is too big (-150552), > clearing out > > > [root@vizioroom105 ~]# uname -a > Linux vizioroom105.msgnet.com 2.6.27.10 #2 SMP Thu Jan 22 09:09:44 EST > 2009 x86_64 x86_64 x86_64 GNU/Linux > > [root@vizioroom105 ~]# arecord -l > **** List of CAPTURE Hardware Devices **** > card 0: AF [VF0560 Live! Cam Optia AF], device 0: USB Audio [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: SB [HDA ATI SB], device 1: ALC883 Digital [ALC883 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: SB [HDA ATI SB], device 2: ALC883 Analog [ALC883 Analog] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > > [root@vizioroom105 ~]# su silentm -c "/usr/bin/vcamera -vvvvv -I dummy > -V dummy --dscp=0x01 v4l2:// :v4l2-dev=/dev/video0 :v4l2-adev=hw.1,0 > :v4l2-width=720 :v4l2-height=480 :v4l2-standard=0 --sout > '#transcode{vcodec=mp4v,vb=6000,fps=30,scale=1,acodec=mpga,ab=32,channels=1}:duplicate{dst=rtp{mux=ts,port=1234,sdp=rtsp://192.168.1.179:8080/test.sdp}}' > vlc://quit" > > Best Regards, > -- > Rob Krakora > Senior Software Engineer > MessageNet Systems > 101 East Carmel Dr. Suite 105 > Carmel, IN 46032 > (317)566-1677 Ext. 206 > (317)663-0808 Fax > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

2 1

[alsa-devel] Advice on ASoC support for series of boards
by Ben Dooks 15 May '09

15 May '09

I would like some advice before I start on a project to move our series of boards to the ASoC framework. All the boards shared the same I2S wiring and all but one have the same TI CODEC wired in similar configuration. The differences are in the presence and connection of a speaker AMP (GPIO controlled). Is there a standard method for binding an AMP to an driver, or is it necessary to have the AMPs in the driver itself? If not, what would people recommend to be the best example of this? -- Ben Q: What's a light-year? A: One-third less calories than a regular year.

2 1

[alsa-devel] How to set TWL4030 to Option2
by Lopez Cruz, Misael 15 May '09

15 May '09

I'm adding the voice dai to sdp3430 machine driver, but the codec's "twl4030_voice_startup" function has a requirement for voice to work: the codec has to be in option2. How can I switch the codec to option2 mode? There is no control for doing that, and the only way I see is to directly modify the reg cache, but I assume that is not the right way to do it. Could you please clarify? -Misa

4 10

Re: [alsa-devel] [PATCH] TXx9: Add ACLC support
by Atsushi Nemoto 15 May '09

15 May '09

On Fri, 15 May 2009 15:45:36 +0200 (CEST), Geert Uytterhoeven <Geert.Uytterhoeven(a)sonycom.com> wrote: > > Add platform support for ACLC of TXx9 SoCs. > > I gave it a try on my RBTX4927, and after some fight^H^H^H^H^Hplaying with the > alsa tools, I could play sound using `speaker-test --test wav --channels 2'! Good news! Thanks for your trial. --- Atsushi Nemoto

1 0

[alsa-devel] Should snd_pcm_drop() interrupt snd_pcm_writei() call?
by Marek Peca 15 May '09

15 May '09

Hello, I would like to know, how should snd_pcm_drop() call influence snd_pcm_writei() being simultaneously executed. I expected, that after issuing snd_pcm_drop() (from another thread), the snd_pcm_write*() in progress should be terminated almost immediately with some return value (error or short write). According to my observation, this seem to work on ALSA lib & driver 1.0.14rc3, snd_ali5451. However, it never or almost never works on another machine with ALSA lib 1.0.16 (Debian package libasound2 [1.0.16-2], driver unknown, sorry). There, the snd_pcm_write*() _never_returns_ after snd_pcm_drop(), and remain stuck forever. I would like to ask you, which behaviour of snd_pcm_drop/snd_pcm_write* is the correct one. I suppose the snd_pcm_write*() should be interrupted, otherwise, an extra thread would be needed in many applications. However, I don't know, why does not it work in the later version of ALSA. I have not found the answer in Doxygen, and since I am real ALSA newbie, I gave up reading ALSA source, especially because I have seen different behaviour among two different versions. Excuse my impatience. Thank you for a clarification. Best regards, Marek.

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[alsa-devel] [PATCH] parisc/harmony: fix printk format warning
by Alexander Beregalov 15 May '09

15 May '09

Fix this warning: sound/parisc/harmony.c:938: warning: format '%lx' expects type 'long unsigned int', but argument 2 has type 'resource_size_t' Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/parisc/harmony.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 6055fd6..63ae0f9 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -935,7 +935,7 @@ snd_harmony_create(struct snd_card *card, h->iobase = ioremap_nocache(padev->hpa.start, HARMONY_SIZE); if (h->iobase == NULL) { printk(KERN_ERR PFX "unable to remap hpa 0x%lx\n", - padev->hpa.start); + (unsigned long)padev->hpa.start); err = -EBUSY; goto free_and_ret; } -- 1.6.2.5

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