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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); }

1 0

[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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[alsa-devel] Infrasonic Quartet
by Vedran Miletić 25 Nov '08

25 Nov '08

Auzen started to distribute one cool new Envy24HT-S card, and it looks really interesting to me. Specs are awesome, and it also has word clock on card! http://www.auzentech.com/site/products/infra_quartet.php http://www.infra-sonic.com/site/p_PCI_quartet.php http://www.newegg.com/Product/Product.aspx?Item=N82E16829156007 According to envy24.svobodno.com, it has: Infrasonic Quartet 0x3030 0x4953 Envy24HT-S - supported by ALSA AK4620B (ADC / DAC) - not sure, probably not supported? AK4112 - not sure either? However it has Xilinx on card in similar way as ESI Juli@ does, so it's probably not a standard Envy24HT-S card. Any ideas if this card could be supported? -- Vedran Miletić

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[alsa-devel] [topic/asoc][RFC 0/1] ASoC: OMAP: Add support for mono link configuration to McBSP DAI
by Jarkko Nikula 25 Nov '08

25 Nov '08

Hi I'm trying to cover also pure mono codecs like bluetooth audio with McBSP DAI driver. Protocol part is trivial like patch here but interfacing with machine driver is somewhat question mark. With this patch, machine driver needs to "hack" with cpu_dai in its init function: cpu_dai->playback.channels_min = 1; cpu_dai->playback.channels_max = 1; cpu_dai->capture.channels_min = 1; cpu_dai->capture.channels_max = 1; Another idea is to introduce new omap_mcbsp_dai_mono[] array in omap-mcbsp.c and initialize those with modified builder macro, e.g: -#define OMAP_MCBSP_DAI_BUILDER(link_id) \ +#define OMAP_MCBSP_DAI_BUILDER(link_id, min_ch, max_ch) \ { \ .name = "omap-mcbsp-dai-"#link_id, \ .id = (link_id), \ .type = SND_SOC_DAI_I2S, \ .playback = { \ - .channels_min = 2, \ - .channels_max = 2, \ + .channels_min = (min_ch), \ + .channels_max = (max_ch), \ ... struct snd_soc_dai omap_mcbsp_dai[] = { - OMAP_MCBSP_DAI_BUILDER(0), - OMAP_MCBSP_DAI_BUILDER(1), + OMAP_MCBSP_DAI_BUILDER(0, 2, 2), + OMAP_MCBSP_DAI_BUILDER(1, 2, 2), #if NUM_LINKS >= 3 ... and +struct snd_soc_dai omap_mcbsp_dai_mono[] = { + OMAP_MCBSP_DAI_BUILDER(0, 1, 1), + OMAP_MCBSP_DAI_BUILDER(1, 1, 1), +#if NUM_LINKS >= 3 ... but this looks a bit overkill if typical configuration has only one such a mono link and might confuse if a machine driver still needs to reference those arrays using sequential array index like: .cpu_dai = &omap_mcbsp_dai[0], and .cpu_dai = &omap_mcbsp_dai_mono[1], sequential since if I like that cpu_dai->id is also sequential for links. Opinnions or better ideas :-) -- Jarkko

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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

1 0

[alsa-devel] Medion MD96420. ALC883/ALC268 conflicts
by Nigel Henry 25 Nov '08

25 Nov '08

I've been helping GN out on the user list, to get his sound working. The sound is now working, but only by using the options line below. options snd-hda-intel probe_mask=1. I see on the changelog for 1.0.18 > 1.0.18a that Takashi added a quirk for a Medion MD96630 as below. <quote> - ALSA: hda - Add a quirk for MEDION MD96630 Use model=lenovo-ms7195-dig for MEDION MD96630 laptop (17c0:4085) with ALC888 codec. Reference: Novell bnc#412548 https://bugzilla.novell.com/show_bug.cgi?id=412528 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> <end quote> Some details of his machine, OS, and other output below. Medion MD96420. Model MIM 2300 Notebook PC OS: Fedora 9, with alsadriver 1.0.16 upgraded to 1.0.18a Desktop: KDE 4.1 /sbin/lspci (soundcard part) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Mitac Unknown device 8227 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fc200000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel [graeme@flintstone ~]$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc200000 irq 22 [graeme@flintstone ~]$ grep ^Codec /proc/asound/card?/codec* /proc/asound/card0/codec#0:Codec: Realtek ALC883 /proc/asound/card0/codec#1:Codec: LSI ID 1040 /proc/asound/card0/codec#2:Codec: Realtek ALC268 A bit of info from Graeme (aka GN), after asking him to try probe_mask=1. <quote> I tried the different probe-mask=?. Probe-mask=1 works OK but not the others. model=lenovo-ms7195-dig doesn't work. Kmix shows varying channels depending on the 'model' or 'probe-mask' used. probe-mask=1 gives the most channels. alsamixer now loads up OK but still no sound. Shows as ALC883 with 'probe-mask=1' Starting Amarok elicits the xine error dialogue window with "xine was unable to initialize any audio drivers" UPDATE: With 'probe-mask=1 and using Rhythmbox I can get sound out of the speakers and the front headphone jack. Volume not very high out of the speakers but is good from the headphones. Too much system activity or keyboard activity (I'm writing this listening to Tchaikovsky and I can hear interruptions from the keyboard at times) seems to interfere with the audio stream. Also there is a lot of noise from the CD drive coming through the earphones. Don't know how that is happening. Pulseaudio Vol Cntl can be used as can the panel Vol. Cntl. and also alsamixer. <end quote> The last things I asked him to try, were to disable Pulseaudio, by removing the package alsa-plugins-pulseaudio, and to try the latest snapshot for the alsa driver. Disabling pulseaudio has resulted in good sound levels from the speakers, but as he is away for a few days, he hasn't yet tried the latest snapshot of the alsa driver. A few extra details. Without a model option all snd modules were loaded, but no card was detected. Using the option model=medion, resulted in the card being detected, but problems with alsamixer as below. model=mitac gives same results as below. <quote from Graeme's reply on user list> [graeme@flintstone ~]$ amixer ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,2,0) appears twice or more amixer: Mixer default load error: Invalid argument [graeme@flintstone ~]$ alsamixer ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,2,0) appears twice or more alsamixer: function snd_mixer_load failed: Invalid argument <end quote> Then I searched through the changelog, found the bug report about the codec conflicts, and suggested probe_mask=1, which has got the sound working. I think I've gone about as far as I can now, which is why I'm posting this to the devel list. Nigel.

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[alsa-devel] Advice requested for developing C01U support via snd-usb-audio
by Keith A. Milner 24 Nov '08

24 Nov '08

Hi, New to this mailing list so please go easy on me. I've been reading through the Alsa docs, wiki, and still not sure of the best approach, so looking for advice from the experts. I'm looking to develop support for the Samson C01U USB studio microphone. http://www.samsontech.com/products/productpage.cfm?prodID=1810 This currently does work via the snd-usb-audio alsa driver, but not in a way that is particularly usable (to the extent that many users have been reporting it as "not working". I thought I might attempt to remedy this. The problem is that, internally, the device contains two digitally controlled amplifiers. Both of these gain controls are exposed via the USB interface (and, hence, by snd-usb-audio) and you can even access the audio stream which is output from each amp. The problem is the raw gain controls are not meant to be accessed directly in this way. The output of one of the amps is fed into the input of the other. With the Windows and Mac OSX drivers, only one gain control is presented. When this is is altered, the driver adjusts the gain of both amplifiers in order to get the correct output level but with minimum noise. In effect, there is a "virtual gain control" which is used to adjust the real gain controls in a predefined way. By careful "twiddling" of the controls it's possible to get the microphone working properly, but it is, quite frankly, a very poor way to handle the problem and is really beyond what should be expected of most users. I would like to do something about this if I can. My aim is to develop something which presents a single gain control to users which controls the internal amp gains in a sensible way. The question is, what is the best way to approach this? Looking at it, I believe there are three possible approaches. In order of my preference they are: 1) Develop at alsa driver level to get a driver which presents a single mixer control to users 2) Develop a new plugin for alsa which can be patched in via asoundrc to provide a generic "Virtual gain control" 3) Provide a user-space tool, a bit like envy24control, which provides this capability My question to the alsa-devel community is which path should I take? As I said my preference is to do something at the driver level simply because it would be most transparent to users and applications: it would simply work. It may also require less code. If this is the correct path to take, this then presents some sub-options (again in order of preference): a) Patch the existing snd-usb-audio driver with some vendor-specific code for this devices (and others which will be similar, like the C03U) b) Fork the snd-usb-driver into a new vendor-specific version and modify as before I prefer a) again because it is transparent to the users, whilst b) will require modprobe or udev configuration, but there may be reasons why it is undesirable to "bloat" the code of snd-usb-audio with vendor-specific tweaks. Bear in mind at this stage I haven't even looked at any of the code yet. Once I do I'm sure to have more questions. But at the moment my question is simply: which is the best way to approach this to be compatible with the aims and philosophy of Alsa? Many thanks, -- Keith A. Milner

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[alsa-devel] [PATCH] ASoC: TWL4030: Disable soft-volume
by Peter Ujfalusi 24 Nov '08

24 Nov '08

Keep Soft-volume disabled for now, since if it is enabled the FGAIN volume controls are not working in the current configuration: CODEC_MODE:OPT_MODE = 1 OPTION:ARXR2_EN = 1 OPTION:ARXL2_EN = 1 OPTION:ARXR1_EN = 0 OPTION:ARXL1_VRX_EN = 0 RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1) RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1) After the patch, FGAIN volume control works. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3348951..49c7da3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -87,7 +87,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ 0x00, /* REG_BOOST_CTL (0x2E) */ - 0x01, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_SOFTVOL_CTL (0x2F) */ 0x00, /* REG_DTMF_FREQSEL (0x30) */ 0x00, /* REG_DTMF_TONEXT1H (0x31) */ 0x00, /* REG_DTMF_TONEXT1L (0x32) */ -- 1.6.0.3

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[alsa-devel] [PATCH] Remove unused variable from tlv320aic23 codec
by Richard A. Holden III 24 Nov '08

24 Nov '08

Remove codec var from tlv320aic23_set_dai_sysclk, it is assigned to but never read. Signed-off-by: Richard A. Holden III <aciddeath(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/tlv320aic23.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308da..45395d0 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -421,8 +421,6 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - switch (freq) { case 12000000: return 0; -- 1.5.6.4

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