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[alsa-devel] Getting the correct capture mixer
by Ludovico Cavedon 29 Apr '09

29 Apr '09

Hi, I am working on fixing some audio related bugs of a VoIP application (qutecom) which, like most VoIP applications, allows the user to change the playback and recording volume. For playback volume I am opening the "Master" mixer element. However I am not able to figure out how the application should get the correct mixer for playback. Is there any rule? For example, my sound card has two capture volumes, one "Int Mic" and one "Ext Mic". I guess I should choose on the basis of what is the user-selected mic. In fact there are also two mutually exclusive switches "IntMic" and "ExtMic". But how should the application understand that? Here part of the output of amixer, my sound card is "Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]" ---------------------------- Simple mixer control 'Ext Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'ExtMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Int Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'IntMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] ------------------------ As an aside, it looks like that the capture volume has no effect, as well all as the cswitches and pswitches of "Int Mic" and "Ext Mic". Thanks, Ludovico

1 0

[alsa-devel] No sound on Acer Aspire 6935G
by Karthik Ramgopal 29 Apr '09

29 Apr '09

Hi, I am using Ubuntu 9.04 Jaunty Jackalope with the latest alsa-driver snapshot provided by Takashi. The sound system does not work without a hda-verb command manually executed every time the system starts up. The command I execute is: hda-verb /dev/snd/hwC0D0 0x15 SET_EAPD_BTLENABLE 2 If I do this I get audio out. But my internal mic does not work and nor does jack sensing. When I insert a device into the headphone jack the speakers do not get muted and there is also no sound from the headphones themselves. Even an external mic/line-in device does not work. I have attached the output of alsa-info.sh here.. Please go through it and try to solve the problem. Thanks and Regards, Karthik

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[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 29 Apr '09

29 Apr '09

The following changes since commit 8a1f936acdfd53cb0a981f3f80483863dcd84fa9: Peter Ujfalusi (1): ASoC: TWL4030: Add 4 channel TDM support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Daniel Mack (3): ASoC: add SOC_DOUBLE_EXT macro ASoC: cs4270: fix Master Capture Switch polarity ASoC: cs4270: add Master Playback Switch Jon Smirl (3): ASoC: Set the MPC5200 i2s driver to BROKEN status. ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s ASoC: Rename the PSC functions to DMA Jonathan Cameron (1): ASoC WM8940 Driver Joonyoung Shim (1): ASoC: TWL4030: Fix gain control for earpiece amplifier Mark Brown (10): ASoC: Fix S3C64xx IIS device registration and support both ports ASoC: S3C2412: Failing to get the I2S clock is an error ASoC: Enforce symmetric rates for S3C64xx I2S interface ASoC: Include WM8350 register definitions in CODEC header ASoC: s3c-i2s-v2 diagnostic improvements ASoC: Use our registration function for S3C64xx ASoC: Fix logic in WM8350 master clocking check Merge branch 'for-2.6.30' into for-2.6.31 ASoC: Staticise TLV values in WM8940 Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (1): ASoC: Beagle: Add support for 4 channel include/sound/soc.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 44 ++- sound/soc/codecs/twl4030.c | 8 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8350.h | 1 + sound/soc/codecs/wm8940.c | 955 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8940.h | 104 +++++ sound/soc/fsl/Kconfig | 6 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/mpc5200_dma.c | 457 +++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 81 ++++ sound/soc/fsl/mpc5200_psc_i2s.c | 644 ++++----------------------- sound/soc/omap/omap3beagle.c | 26 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 148 ++++-- sound/soc/s3c24xx/s3c64xx-i2s.h | 2 +- 18 files changed, 1867 insertions(+), 629 deletions(-) create mode 100644 sound/soc/codecs/wm8940.c create mode 100644 sound/soc/codecs/wm8940.h create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h

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[alsa-devel] Setting the AC97 private data
by Jon Smirl 28 Apr '09

28 Apr '09

My AC97 CPU read/write routines need private data set so that they know which port to use. The mpc5200 has multiple PSCs that can function in AC97 mode. static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { ... if (rtd->socdev->card->codec->ac97) rtd->socdev->card->codec->ac97->private_data = psc_dma; .. } Is there some easier way to set this data? -- Jon Smirl jonsmirl(a)gmail.com

2 3

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 28 Apr '09

28 Apr '09

The following changes since commit 0c95de73a711d376dc17afe484f919bd5b66c016: Jon Smirl (1): ASoC: Set the MPC5200 i2s driver to BROKEN status. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Joonyoung Shim (1): ASoC: TWL4030: Fix gain control for earpiece amplifier sound/soc/codecs/twl4030.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-)

1 0

[alsa-devel] [PATCH] ALSA: au1x00: convert to platform device
by omelinux.net＠alsa-project.org 28 Apr '09

28 Apr '09

Make sound/mips/au1x00.c a proper platform_driver. Cc: Charles Eidsness <charles(a)cooper-street.com> Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- Compile-tested only; I don't have the right hardware to test this. Charles: If you still have access to the hardware, could you please test this patch? Just add something like the following to your board's platform device registration: #include <asm/mach-au1x00/au1000_dma.h> static struct resource au1xxx_ac97c_res[] = { [0] = { .start = AC97_PHYS_ADDR, .end = AC97_PHYS_ADDR + 0xfff, .flags = IORESOURCE_MEM, }, [1] = { .start = AU1000_AC97C_INT, .end = AU1000_AC97C_INT, .flags = IORESOURCE_IRQ, }, [2] = { .start = DMA_ID_AC97C_TX, .end = DMA_ID_AC97C_TX, .flags = IORESOURCE_DMA, }, [3] = { .start = DMA_ID_AC97C_RX, .end = DMA_ID_AC97C_RX, .flags = IORESOURCE_DMA, }, }; static struct platform_device au1xxx_ac97c_device = { .name = "au1000-ac97c", .id = -1, .resource = au1xxx_ac97c_res, .num_resources = ARRAY_SIZE(au1xxx_ac97c_res), }; and register the au1xxx_ac97c_device. Thank you! sound/mips/au1x00.c | 244 +++++++++++++++++++++++++++++++-------------------- 1 files changed, 147 insertions(+), 97 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 3e763d6..11b5865 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -37,6 +37,7 @@ #include <linux/ioport.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/initval.h> @@ -97,6 +98,7 @@ struct snd_au1000 { struct snd_pcm *pcm; struct audio_stream *stream[2]; /* playback & capture */ + int dmaid[2]; /* tx(0)/rx(1) DMA ids */ }; /*--------------------------- Local Functions --------------------------------*/ @@ -464,13 +466,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000) spin_lock_init(&au1000->stream[CAPTURE]->dma_lock); flags = claim_dma_lock(); - if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX, + if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(au1000->dmaid[0], "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED, au1000->stream[PLAYBACK])) < 0) { release_dma_lock(flags); return -EBUSY; } - if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX, + if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(au1000->dmaid[1], "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED, au1000->stream[CAPTURE])) < 0){ release_dma_lock(flags); @@ -550,69 +552,12 @@ get the interrupt driven case to work efficiently */ spin_unlock(&au1000->ac97_lock); } -static int __devinit -snd_au1000_ac97_new(struct snd_au1000 *au1000) -{ - int err; - struct snd_ac97_bus *pbus; - struct snd_ac97_template ac97; - static struct snd_ac97_bus_ops ops = { - .write = snd_au1000_ac97_write, - .read = snd_au1000_ac97_read, - }; - - if ((au1000->ac97_res_port = request_mem_region(CPHYSADDR(AC97C_CONFIG), - 0x100000, "Au1x00 AC97")) == NULL) { - snd_printk(KERN_ERR "ALSA AC97: can't grap AC97 port\n"); - return -EBUSY; - } - au1000->ac97_ioport = (struct au1000_ac97_reg *) - KSEG1ADDR(au1000->ac97_res_port->start); - - spin_lock_init(&au1000->ac97_lock); - - /* configure pins for AC'97 - TODO: move to board_setup.c */ - au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); - - /* Initialise Au1000's AC'97 Control Block */ - au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; - udelay(10); - au1000->ac97_ioport->cntrl = AC97C_CE; - udelay(10); - - /* Initialise External CODEC -- cold reset */ - au1000->ac97_ioport->config = AC97C_RESET; - udelay(10); - au1000->ac97_ioport->config = 0x0; - mdelay(5); - - /* Initialise AC97 middle-layer */ - if ((err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus)) < 0) - return err; - - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = au1000; - if ((err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97)) < 0) - return err; - - return 0; -} - /*------------------------------ Setup / Destroy ----------------------------*/ -void -snd_au1000_free(struct snd_card *card) +static void snd_au1000_free(struct snd_card *card) { struct snd_au1000 *au1000 = card->private_data; - if (au1000->ac97_res_port) { - /* put internal AC97 block into reset */ - au1000->ac97_ioport->cntrl = AC97C_RS; - au1000->ac97_ioport = NULL; - release_and_free_resource(au1000->ac97_res_port); - } - if (au1000->stream[PLAYBACK]) { if (au1000->stream[PLAYBACK]->dma >= 0) free_au1000_dma(au1000->stream[PLAYBACK]->dma); @@ -624,71 +569,176 @@ snd_au1000_free(struct snd_card *card) free_au1000_dma(au1000->stream[CAPTURE]->dma); kfree(au1000->stream[CAPTURE]); } -} + if (au1000->ac97_res_port) { + /* put internal AC97 block into reset */ + au1000->ac97_ioport->cntrl = AC97C_RS; + iounmap(au1000->ac97_ioport); + au1000->ac97_ioport = NULL; + release_and_free_resource(au1000->ac97_res_port); + au1000->ac97_res_port = NULL; + } +} -static struct snd_card *au1000_card; +static struct snd_ac97_bus_ops ops = { + .write = snd_au1000_ac97_write, + .read = snd_au1000_ac97_read, +}; -static int __init -au1000_init(void) +static int __devinit au1000_ac97_probe(struct platform_device *pdev) { int err; + void __iomem *io; + struct resource *r; struct snd_card *card; struct snd_au1000 *au1000; + struct snd_ac97_bus *pbus; + struct snd_ac97_template ac97; err = snd_card_create(-1, "AC97", THIS_MODULE, - sizeof(struct snd_au1000), &card); + sizeof(struct snd_au1000), &card); if (err < 0) - return err; + goto out0; - card->private_free = snd_au1000_free; au1000 = card->private_data; au1000->card = card; - au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - au1000->stream[CAPTURE ] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - /* so that snd_au1000_free will work as intended */ - au1000->ac97_res_port = NULL; - if (au1000->stream[PLAYBACK]) - au1000->stream[PLAYBACK]->dma = -1; - if (au1000->stream[CAPTURE ]) - au1000->stream[CAPTURE ]->dma = -1; - - if (au1000->stream[PLAYBACK] == NULL || - au1000->stream[CAPTURE ] == NULL) { - snd_card_free(card); - return -ENOMEM; + /* TX DMA ID */ + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) { + err = -ENODEV; + snd_printk(KERN_INFO "no TX DMA platform resource!\n"); + goto out1; } - - if ((err = snd_au1000_ac97_new(au1000)) < 0 ) { - snd_card_free(card); - return err; + au1000->dmaid[0] = r->start; + + /* RX DMA ID */ + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) { + err = -ENODEV; + snd_printk(KERN_INFO "no RX DMA platform resource!\n"); + goto out1; } + au1000->dmaid[1] = r->start; - if ((err = snd_au1000_pcm_new(au1000)) < 0) { - snd_card_free(card); - return err; + au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); + if (!au1000->stream[PLAYBACK]) + goto out1; + au1000->stream[PLAYBACK]->dma = -1; + + au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); + if (!au1000->stream[CAPTURE]) + goto out2; + au1000->stream[CAPTURE]->dma = -1; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) + goto out3; + + err = -EBUSY; + au1000->ac97_res_port = request_mem_region(r->start, + r->end - r->start + 1, pdev->name); + if (!au1000->ac97_res_port) { + snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); + goto out3; } + io = ioremap(r->start, r->end - r->start + 1); + if (!io) + goto out4; + + au1000->ac97_ioport = (struct au1000_ac97_reg *)io; + + spin_lock_init(&au1000->ac97_lock); + + /* configure pins for AC'97 + TODO: move to board_setup.c */ + au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); + + /* Initialise Au1000's AC'97 Control Block */ + au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; + udelay(10); + au1000->ac97_ioport->cntrl = AC97C_CE; + udelay(10); + + /* Initialise External CODEC -- cold reset */ + au1000->ac97_ioport->config = AC97C_RESET; + udelay(10); + au1000->ac97_ioport->config = 0x0; + mdelay(5); + + /* Initialise AC97 middle-layer */ + err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus); + if (err < 0) + goto out5; + + /* from here on let ALSA call the special freeing function */ + card->private_free = snd_au1000_free; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = au1000; + err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97); + if (err < 0) + goto out1; + + err = snd_au1000_pcm_new(au1000); + if (err < 0) + goto out1; + strcpy(card->driver, "Au1000-AC97"); strcpy(card->shortname, "AMD Au1000-AC97"); sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver"); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto out1; + + printk( KERN_INFO "ALSA AC97: Driver Initialized\n" ); + + platform_set_drvdata(pdev, card); - printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); - au1000_card = card; return 0; + +out5: + au1000->ac97_ioport->cntrl = AC97C_RS; + iounmap(io); +out4: + release_and_free_resource(au1000->ac97_res_port); +out3: + kfree(au1000->stream[PLAYBACK]); +out2: + kfree(au1000->stream[PLAYBACK]); +out1: + snd_card_free(card); +out0: + return err; } -static void __exit au1000_exit(void) +static int __devexit au1000_ac97_remove(struct platform_device *pdev) { - snd_card_free(au1000_card); + struct snd_card *card = platform_get_drvdata(pdev); + snd_card_free(card); + return 0; } -module_init(au1000_init); -module_exit(au1000_exit); +struct platform_driver au1000_ac97c_driver = { + .driver = { + .name = "au1000-ac97c", + .owner = THIS_MODULE, + }, + .probe = au1000_ac97_probe, + .remove = __devexit_p(au1000_ac97_remove), +}; + +static int __init au1000_ac97_load(void) +{ + return platform_driver_register(&au1000_ac97c_driver); +} + +static void __exit au1000_ac97_unload(void) +{ + platform_driver_unregister(&au1000_ac97c_driver); +} +module_init(au1000_ac97_load); +module_exit(au1000_ac97_unload); -- 1.6.2

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[alsa-devel] File SUPPORTED_KERNELS should be updated
by Susan Cragin 28 Apr '09

28 Apr '09

The following is the entire text of that file. ------------------------------------------------------------------------ The alsa-drivers in this package are designed for the following kernels: - Vanilla 2.6.25 or earlier - Vanilla 2.4.31 or earlier - Vanilla 2.2.26 or earlier It's not guaranteed that they work with any newer version than above or modified kernels by distributors. ----------------------------------------------------------------------- Isn't the information obsolete? Aren't the new alsa-drivers in fact destined for 2.6.28 and 2.6.29? Or am I missing something? Susan

2 1

Re: [alsa-devel] No sound with Creative Labs CA0106 Soundblaster
by Takashi Iwai 28 Apr '09

28 Apr '09

At Thu, 16 Apr 2009 23:23:55 +0200, Bernhard Posselt wrote: > > Am Donnerstag 16 April 2009 16:10:29 schrieben Sie: > > At Wed, 15 Apr 2009 04:00:26 +0200, > > > > Bernhard Posselt wrote: > > > Hi, some months ago i bought a new soundcard for my PC which costed 20€. > > > This card is not working. > > > > > > Im using Kubuntu Jaunty Beta, its a fresh install from the beta iso. I > > > also upgraded the alsadrivers with this package > > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver- > > >snapshot.tar.bz2 > > > > > > Still no sound > > > > Please show your alsa-info.sh output (run with --no-upload option). > > > > If it's an unknown model, try to pass some serial number via subsystem > > module option, such as, > > > > # modprobe snd-ca0106 subsystem=0x10131102 > > > > The available numbers are found in ca0106_chip_details[] list in > > ca0106/ca0106_main.c. > > > > > > Takashi > > It's an Audigy Soundblaster LE, or better was. I guess you thought that i > would be using the SE model. No, I just thought you are using a model that isn't listed in the ca0106 driver's white list, no matter which one. Createive tends to ship quite lots of variants with different SSIDs although many are compatible. Takashi

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix gain control for earpiece amplifier
by Joonyoung Shim 28 Apr '09

28 Apr '09

The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a7875c2..bf522a9 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -785,6 +785,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); /* + * Gain control for earpiece amplifier + * 0 dB to 12 dB in 6 dB steps (mute instead of -6) + */ +static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1); + +/* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps */ @@ -859,7 +865,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 4, 3, 0, output_tvl), SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", -- 1.6.0.4

2 1

[alsa-devel] [PATCH 1/1] indigo-express: add missing 64KHz flags
by Giuliano Pochini 28 Apr '09

28 Apr '09

Indigo-express cards also support 64KHz sampling rate: this patch adds missing SNDRV_PCM_RATE_64000 flags. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-kernel/pci/echoaudio/indigodjx.c__orig 2009-04-27 22:57:20.000000000 +0200 +++ alsa-kernel/pci/echoaudio/indigodjx.c 2009-04-27 22:58:04.000000000 +0200 @@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardw .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, --- alsa-kernel/pci/echoaudio/indigoiox.c__orig 2009-04-27 22:56:46.000000000 +0200 +++ alsa-kernel/pci/echoaudio/indigoiox.c 2009-04-27 22:57:54.000000000 +0200 @@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardw .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, -- Giuliano.

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