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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] 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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[alsa-devel] (no subject)
by Nitin Mahajan 24 Nov '08

24 Nov '08

unsubscribe Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/

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[alsa-devel] More constants for alsa-lib
by flameeyes＠gmail.com 24 Nov '08

24 Nov '08

git://git.flameeyes.eu/others/alsa/alsa-lib.git flameeyes Diego E. 'Flameeyes' Pettenò (6): Make seq, rawmidi and control operation structures static const. Make snd_pcm_hw_params_names static to pcm_params.c . Make all the PCM plugins ops structure constant. Make all the remaining ops structure constants. Make string arrays as constant as possible. Mark static tables as constant when possible. (This time, signed off right away ;)) -- Diego "Flameeyes" Pettenò http://blog.flameeyes.eu/

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[alsa-devel] [PATCH] Added Realtek ALC888 model entry for the Acer Aspire 4930G laptop
by Petry Vincent 24 Nov '08

24 Nov '08

Hello, I would like to contribute a patch that adds a model entry for the Acer Aspire 4930G laptop. Its chipset is a Realtek ALC888S-VC. The new model is based on the "acer" model and fixes the following bugs missing features: - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4223 Reversed stereo (invalid report, please close it) - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4224 Headphone jack sense - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4225 Internal mic - channel mode (it was missing from the base acer model) The model name is "acer-aspire-4930g" but could be renamed to something more general, if necessary. See attached patch file. It has been made against alsa-driver-1.0.18a. Thanks, Vincent Petry

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[alsa-devel] Suggestion: Jack nodes detection
by Vincent Petry 24 Nov '08

24 Nov '08

Hi, I have noticed that from one model to another in ALC888 (maybe others as well), the nodes for Int speakers, HP out, are not always the same. But in the current driver model in patch_realtek.c we always hard-code the node numbers, for example 0x14 for int speakers, 0x15 for hp out, etc. It would be good to have it this way: 1. Add a spec attribute that contains the output nids in a given order or a map, like this for my laptop: alc888_acer_aspire_jack_nids = { {0x14, NID_INT_SPEAKER}, {0x15, NID_HP_OUT}, {0x18, NID_MIC_IN}, {0x1a, NID_LINE_IN}, {} }; For another laptop, that has built-in subwoofer: alc888_fujitsu_xa2530_jack_nids = { { {0x14, NID_INT_SPEAKER}, {0x15, NID_INT_SPEAKER}, // subwoofer, also considered as int speaker for the jack sense muting {0x17, NID_LINE_OUT}, // this is also side {0x18, NID_MIC_IN}, {0x1a, NID_LINE_IN}, {} }; This way we could then use a function to access the NID number of each one of them, for example get_jack_nid(..., NID_INT_SPEAKER). 2. Once this is implemented, we could add a detection routine that would build the above map automatically, based on the spec info returned by the codec. 3. Then I think we should add some special "detection" model that has no hard-coded values that anyone could use by default, if no suitable model def exists. How about this idea? Is someone already working on it? Thanks, Vincent Petry

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[alsa-devel] [PATCH 0/7] minor HDMI cleanups and fixes
by Wu Fengguang 24 Nov '08

24 Nov '08

Takashi, Here are another set of HDMI cleanups and fixes. [PATCH 1/7] hda - minor HDMI code cleanups [PATCH 2/7] hda - report selected CA index for Audio InfoFrame [PATCH 3/7] hda - make HDMI messages more user friendly [PATCH 4/7] hda - ELD proc interface write updates [PATCH 5/7] hda - document the ELD proc interface [PATCH 6/7] hda - fix DisplayPort naming [PATCH 7/7] hda - fix build warning when CONFIG_PROC_FS=n Thanks, Fengguang --

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