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[alsa-devel] [PATCH 2/2] sound/soc/soc-core.c: cleanup kernel-doc
by Qinghuang Feng 25 Nov '08

25 Nov '08

There is no argument named @clk_id in snd_soc_dai_set_fmt, remove its' comment. Signed-off-by: Qinghuang Feng <qhfeng.kernel(a)gmail.com> --- diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16c7453..81ee4ba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1805,7 +1805,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @clk_id: DAI specific clock ID * @fmt: SND_SOC_DAIFMT_ format value. * * Configures the DAI hardware format and clocking.

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[alsa-devel] [PATCH] [ASOC] tosa: move gpio probing to machine callbacks
by Dmitry Baryshkov 25 Nov '08

25 Nov '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/tosa.c | 30 ++++++++++++++++++++++-------- 1 files changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index afefe41..69ca79b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -230,10 +230,32 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; +static int tosa_probe(struct platform_device *dev) +{ + int ret; + + ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0); + if (ret) + gpio_free(TOSA_GPIO_L_MUTE); + + return ret; +} + +static int tosa_remove(struct platform_device *dev) +{ + gpio_free(TOSA_GPIO_L_MUTE); + return 0; +} + static struct snd_soc_machine tosa = { .name = "Tosa", .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + .probe = tosa_probe, + .remove = tosa_remove, }; static struct snd_soc_device tosa_snd_devdata = { @@ -251,11 +273,6 @@ static int __init tosa_init(void) if (!machine_is_tosa()) return -ENODEV; - ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); - if (ret) - return ret; - gpio_direction_output(TOSA_GPIO_L_MUTE, 0); - tosa_snd_device = platform_device_alloc("soc-audio", -1); if (!tosa_snd_device) { ret = -ENOMEM; @@ -272,15 +289,12 @@ static int __init tosa_init(void) platform_device_put(tosa_snd_device); err_alloc: - gpio_free(TOSA_GPIO_L_MUTE); - return ret; } static void __exit tosa_exit(void) { platform_device_unregister(tosa_snd_device); - gpio_free(TOSA_GPIO_L_MUTE); } module_init(tosa_init); -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: SoC: Add support for TI SDP3430
by Misael Lopez Cruz 25 Nov '08

25 Nov '08

This patch add ASoC support for TI SDP3430. It's based on Gumstix Overo SoC code by Steve Sakoman. Signed-off-by: Misael Lopez Cruz <mesak82(a)gmail.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 152 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/sdp3430.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0daeee4..e816fa4 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,4 +38,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_SDP3430 + tristate "SoC Audio support for Texas Instruments SDP3430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 4bae404..64e560b 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o +snd-soc-sdp3430-objs := sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c new file mode 100644 index 0000000..85fd160 --- /dev/null +++ b/sound/soc/omap/sdp3430.c @@ -0,0 +1,152 @@ +/* + * sdp3430.c -- SoC audio for TI OMAP3430 SDP + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * Based on: + * Author: Steve Sakoman <steve(a)sakoman.com> + * + * 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 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int sdp3430_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_ops = { + .hw_params = sdp3430_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp3430_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &sdp3430_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sdp3430 = { + .name = "SDP3430", + .dai_link = &sdp3430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp3430_snd_devdata = { + .machine = &snd_soc_machine_sdp3430, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *sdp3430_snd_device; + +static int __init sdp3430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_3430sdp()) { + pr_debug("Not SDP3430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP3430 SoC init\n"); + + sdp3430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp3430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); + sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; + *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(sdp3430_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp3430_snd_device); + + return ret; +} +module_init(sdp3430_soc_init); + +static void __exit sdp3430_soc_exit(void) +{ + platform_device_unregister(sdp3430_snd_device); +} +module_exit(sdp3430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP3430"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

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[alsa-devel] [topic/asoc][PATCH 0/2] ASoC: OMAP: Add support for mono audio links in McBSP DAI (take #2)
by Jarkko Nikula 25 Nov '08

25 Nov '08

Hi Second version adding support for mono audio links in OMAP McBSP DAI. At this time, the channels_min is defined to 1 in omap-mcbsp.c and no hacks in machine drivers wanting to use mono link are required. If channel restrictions are required, then machine driver can use snd_pcm_hw_constraint_minmax. Now only N810 machine driver is affected since all other OMAP machine drivers are using TWL4030 or TLV320AIC23 codecs which all claim to support only 2 channels so the link remains always stereo even playing mono audio. -- Jarkko

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[alsa-devel] pcm_prepare, and pcm_trigger
by John Kacur 25 Nov '08

25 Nov '08

More ASoC fundamentals questions. when doing cat 8k8bitpcm.wav > /dev/dsp I'm seeing neither myplatform_pcm_prepare, nor myplatform_pcm_trigger called. Is this normal? Who calls these functions? Thanks

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[alsa-devel] problem with scheduling alsa events
by nescivi 25 Nov '08

25 Nov '08

Hiho, In the Midi output code of this opensource project I am working on, there is a problem with the timing when alsa events are scheduled in the future. The events take much longer to arrive (sometimes almost minutes later!). Below is the code snippet. Can anyone see what is wrong in here? Let me know if you need more code... This is with libasound2 1.0.16-2 (Debian unstable). sincerely, Marije int SC_AlsaMidiClient::sendEvent(int outputIndex, int uid, snd_seq_event_t* evt, float late) { snd_seq_real_time time; if ((outputIndex < 0) || (outputIndex >= mNumOutPorts)) return errIndexOutOfRange; snd_seq_ev_set_source(evt, mOutPorts[outputIndex]); if (uid == 0) { // send to all subscribed ports snd_seq_ev_set_subs(evt); } else { // send to specific port int cid, pid; SC_AlsaParseUID(uid, cid, pid); snd_seq_ev_set_dest(evt, cid, pid); } // late is the latency value in seconds in floats if (late > 0.f) { // old time calculation: // time.tv_sec = (unsigned)(late * 1.0e-6f); // time.tv_nsec = (unsigned)(late * 1.0e3f); // new time calculation. The old one was not correct time.tv_sec = (unsigned)(late); // seconds time.tv_nsec = (unsigned)(late * 1.0e9f) % 1000000000; // nanoseconds // should these be (long) instead??? } else { time.tv_sec = time.tv_nsec = 0; } post("MIDI (ALSA): sending event, time %i, %i, late %f\n", time.tv_sec, time.tv_nsec, late); snd_seq_ev_schedule_real(evt, mQueue, 1, &time); snd_seq_event_output_direct(mHandle, evt); return errNone; }

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[alsa-devel] MSI 7260 patch
by Страхиња Радић 25 Nov '08

25 Nov '08

Hi, I submitted an issue with alsa-driver 1.0.18rc3 (see https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4229). Raymond advised me to send the patch, so everybody with the same motherboard could get automatic recognition of sound card model. And here it is. I just don't know if I got it right. Are the hexadecimal values ok? SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG), I got this from alsainfo: !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x14627260 Revision Id: 0x100002 No Modem Function Group found So, I guess 0x1462 0x7260 is ok, right? If that is so, I have attached the diff with this message. Cheers, Strahinya -- _ _ ______________________________________________________ /(_____)\ | ____ __ __ __ __ | \_/~ ~\_/ |/ __// /_/ // // / http://www.gnu.org/home.sr.html | (\'_'/) / / / __ /_\ ' / http://www.stallman.org | (o_o) /_/ /_/ /_//____/ http://alas.matf.bg.ac.yu/~mr99164 | |______________________________________________________| --- a/sound/pci/hda/patch_realtek.c 2008-09-09 18:55:25.000000000 +0200 +++ b/sound/pci/hda/patch_realtek.c 2008-11-24 17:56:31.000000000 +0100 @@ -8320,6 +8320,7 @@ SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),

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[alsa-devel] [PATCH] ASoC: Add support for OMAP3 EVM
by Anuj Aggarwal 25 Nov '08

25 Nov '08

This patch adds ALSA SoC support for OMAP3 EVM using TWL4030 audio codec. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 156 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6c56277..c465139 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,5 +38,12 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index f5da3cc..bbb3a77 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..570af55 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_omap3evm = { + .name = "omap3evm", + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .machine = &snd_soc_machine_omap3evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_debug("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPL"); -- 1.5.6

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[alsa-devel] snd-usb-audio: minimum period time as module option?
by Klaus Schulz 25 Nov '08

25 Nov '08

Hi there. I wrote a small patch which introduces an usb-audio module option called uperiod. Changing to lower values will improve sound quality in some cases. Please comment. THX Klaus --- alsa-driver-1.0.18a/sound/usb/usbaudio.c.orig 2008-11-21 19:19:11.000000000 +0100 +++ alsa-driver-1.0.18a/sound/usb/usbaudio.c 2008-11-24 13:40:12.000000000 +0100 @@ -72,6 +72,7 @@ static int nrpacks = 8; /* max. number static int async_unlink = 1; static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/ static int ignore_ctl_error; +static int uperiod = 125; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); @@ -92,6 +93,8 @@ MODULE_PARM_DESC(device_setup, "Specific module_param(ignore_ctl_error, bool, 0444); MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); +module_param(uperiod, int, 0644); +MODULE_PARM_DESC(uperiod, "Max. size of period per URB."); /* * debug the h/w constraints @@ -108,6 +111,8 @@ MODULE_PARM_DESC(ignore_ctl_error, #define MAX_URBS 8 #define SYNC_URBS 4 /* always four urbs for sync */ #define MIN_PACKS_URB 1 /* minimum 1 packet per urb */ +#define MAX_UPERIOD 1000 +#define MIN_UPERIOD 64 struct audioformat { struct list_head list; @@ -1919,7 +1924,7 @@ static int setup_hw_info(struct snd_pcm_ * in the current code assume the 1ms period. */ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 1000 * MIN_PACKS_URB, + uperiod * MIN_PACKS_URB, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); err = check_hw_params_convention(subs); @@ -3767,6 +3772,11 @@ static int __init snd_usb_audio_init(voi printk(KERN_WARNING "invalid nrpacks value.\n"); return -EINVAL; } + + if (uperiod < MIN_UPERIOD || uperiod > MAX_UPERIOD) { + printk(KERN_WARNING "invalid uperiod value.\n"); + return -EINVAL; + } return usb_register(&usb_audio_driver); }

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[alsa-devel] [PATCH 0/9] ASoC: TWL4030: Add mixer controls for outputs
by Peter Ujfalusi 25 Nov '08

25 Nov '08

Hello, I have tried to clean up the changes that I have at the momnent. After the series you will have 'ultimate' control on how the audio is routed to the outputs of TWL. One can select the possible inputs (for the output) on all cahnnels. Also I have added gain (volume) controls, wherever it is possible, but they are working for both channels in case of stereo outputs. I tried to name the controls to match with the output's name, so you can easily tell which pin(s) the control will touch. The capture part is not touched, yet. I have not included the changes regarding to the SND_SOC_DAPM_OUTPUT() things. So you have one (actually two) outputs and you can use the mixer interface to select where the audio actually routed. The next step is to make correct dapm mapping for the TWL, but that will take some time. Please comment, if I'm doing something that I should not. --- Peter Ujfalusi (9): ASoC: TWL4030: Change the Master volume control to TLV ASoC: TWL4030: Add CGAIN volume control ASoC: TWL4030: Add helper function for mux handling. ASoC: TWL4030: Add helper function for output gain controls ASoC: TWL4030: Add PreDriv outupt mux and volume controls ASoC: TWL4030: Add Headset outupt mux and volume controls ASoC: TWL4030: Add Carkit outupt mux and volume controls ASoC: TWL4030: Add Hands-free outupt mux control ASoC: TWL4030: Add Earpiece outupt mux and volume controls sound/soc/codecs/twl4030.c | 298 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 296 insertions(+), 2 deletions(-)

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