October 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/9] ASoC Blackfin supporting
by Bryan Wu 09 Oct '08

09 Oct '08

Hi Mark, After your last review, Cliff and me updated this patch series: - fix coding style issue - use latest ASoC API - split the whole patch into this 9 patches in a patchset Thanks a lot -Bryan

6 24

Re: [alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by The Source 09 Oct '08

09 Oct '08

So, should anyone start to port OSS driver or should we wait for specs?

1 0

[alsa-devel] dv6627el & audio problems
by yelo_3 09 Oct '08

09 Oct '08

Hello, I want to share my problems with you. I'm running ubuntu intrepid, with these packages versions: linux 2.6.27-6-generic alsa 1.0.17.dfsg-2ubuntu1 I have an hp dv6627el with this audio card model: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) and this is my alsa page: http://www.alsa-project.org/db/?f=457f3992bcf0b34b51772a730071ce18deea75b2 I don't use any additional module option other that the one I will specify. I have 2 problems: 1) mute led does not work. if I use the option power_save (for isntance=3) the mute led lights up (red) after 3 seconds of audio inactivity (and after having closed the audio software), even if the volume is not muted. If I keep the audio software open, the led remains "off" (blue). if instead I don't use the option power_save (so power_save=0) the led never lights up. 2) when using headphones, the speakers are normally muted, but if I hit the mute button twice (or use the volume control applet to mute and unmute), the speakers are activated. Thank you very much for you help. Nicolò

1 0

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with debugfs filesystem
by Troy Kisky 08 Oct '08

08 Oct '08

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> Mark, I didn't base it on dapm's pop_time directory because that hasn't made it to my testing kernel yet. I cannot test directly from the alsa git tree. Thanks for the feedback. diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f616b0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -516,6 +516,9 @@ struct snd_soc_device { struct delayed_work delayed_work; struct work_struct deferred_resume_work; void *codec_data; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_root; +#endif }; /* runtime channel data */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..156d11a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -962,10 +962,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +1000,112 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_DEBUG_FS +static int code_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t code_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(devdata, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t code_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = code_reg_open_file, + .read = code_reg_read_file, + .write = code_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *debugfs_codec_reg; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + debugfs_codec_reg = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!debugfs_codec_reg) + goto exit2; + + socdev->debugfs_root = root; + return; +exit2: + debugfs_remove(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove_recursive(socdev->debugfs_root); + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1217,6 +1319,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_codec_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1240,6 +1343,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #endif mutex_lock(&codec->mutex); + soc_cleanup_codec_debugfs(socdev); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; -- 1.5.4.3

2 1

[alsa-devel] [PATCH] ASoC: Check for machine type in GTA01 machine driver
by Mark Brown 08 Oct '08

08 Oct '08

Since there are now multiple OpenMoko platforms it is more important to check that the machine driver is running on the correct system. This was orgininally generated as part of the initial GTA02 machine port. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/neo1973_wm8753.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c602674..87ddfef 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -24,6 +24,7 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include <asm/mach-types.h> #include <asm/hardware/scoop.h> #include <mach/regs-clock.h> #include <mach/regs-gpio.h> @@ -677,6 +678,12 @@ static int __init neo1973_init(void) DBG("Entered %s\n", __func__); + if (!machine_is_neo1973_gta01()) { + printk(KERN_INFO + "Only GTA01 hardware supported by ASoC driver\n"); + return -ENODEV; + } + neo1973_snd_device = platform_device_alloc("soc-audio", -1); if (!neo1973_snd_device) return -ENOMEM; -- 1.5.6.5

2 1

Re: [alsa-devel] AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 08 Oct '08

08 Oct '08

Not yet. Would please share what you have done to make it work ? Also when I tried to strace aplay , it keeps looking for a </share/alsa/cards/tlv320aic23.conf> file which is not there in the alsa-lib package. Any idea what goes in that file ? Regards Jesslyn On Tue, Oct 7, 2008 at 7:16 PM, Arun KS <arunks(a)mistralsolutions.com> wrote: > Jesslyn Abdul Salam wrote: > >> I got it working. I2C_BOARD_INFO macro was wrongly intialized. Instead of >> tlv320aic23 i had written tlvaic23. Thanks to Arun for pointing this error >> >> > Can you hear audio playing? > Because i had done some hack to make it working in 5912osk. > I want to know you are not facing it or not. > > Arun > > Jesslyn Abdul Salam >> >> >> On Tue, Oct 7, 2008 at 5:36 PM, Jesslyn Abdul Salam < >> jesslyn.abdulsalam(a)gmail.com> wrote: >> >> >> >>> ---------- Forwarded message ---------- >>> From: Jesslyn Abdul Salam <jesslyn.abdulsalam(a)gmail.com> >>> Date: Tue, Oct 7, 2008 at 5:36 PM >>> Subject: Re: [alsa-devel] AIC23 SOC Driver on OSK5912 >>> To: Arun KS <arunks(a)mistralsolutions.com> >>> >>> >>> Yes, it is there. but it shows as "tlv320aic23b" instead of "tlv320aic" >>> Will that be the reason ???? >>> >>> >>> >>> >>> On Tue, Oct 7, 2008 at 5:32 PM, Arun KS <arunks(a)mistralsolutions.com >>> >wrote: >>> >>> >>> >>>> On Tue, Oct 7, 2008 at 5:24 PM, Mark Brown <broonie(a)sirena.org.uk> >>>> wrote: >>>> >>>> >>>>> On Tue, Oct 07, 2008 at 05:18:55PM +0530, Jesslyn Abdul Salam wrote: >>>>> >>>>> >>>>> >>>>>> I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for >>>>>> the >>>>>> >>>>>> >>>>> past >>>> >>>> >>>>> few days. I got the driver patch from the linux-omap/alsa-devel mailing >>>>>> lists. The driver got compiled along just fine. But when the kernel >>>>>> >>>>>> >>>>> boots, >>>> >>>> >>>>> no sound cards have been listed. >>>>>> >>>>>> >>>>> CCing in Arun KS <arunks(a)mistralsolutions.com> who contributed that >>>>> support. >>>>> >>>>> >>>>> >>>>>> Advanced Linux Sound Architecture Driver Version 1.0.17. >>>>>> ASoC version >>>>>> 0.13.2 >>>>>> AIC23 Audio Codec >>>>>> 0.1 >>>>>> MCLK = 12000000 [12000000], usecount = >>>>>> 0 >>>>>> ALSA device >>>>>> list: >>>>>> No soundcards >>>>>> found. >>>>>> Any one have tested this before ? >>>>>> >>>>>> >>>>> Can you check i2c? >>>> >>>> # cat /sys/bus/i2c/devices/1-001b/name >>>> tlv320aic23 >>>> >>>> >>>> >>>>> _______________________________________________ >>>>> Alsa-devel mailing list >>>>> Alsa-devel(a)alsa-project.org >>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>> >>>>> >>>>> >>>> >>> >>> >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >> >> >> > > > Please do not print this email unless it is absolutely necessary. Spread > environmental awareness. > > > -------------------------------------------------------DISCLAIMER------------------------------------------------------ > The information transmitted herewith is confidential and proprietary > information intended only for use by the individual or entity to which it is > addressed. If the reader of this message is not the intended recipient, you > are hereby notified that any review, retransmission, dissemination, > distribution, copying or other use of, or taking of any action in reliance > upon this information is strictly prohibited. If you have received this > communication in error, please contact the sender and delete the material > from your computer. > > -------------------------------------------------------------------------------------------------------------------------------- > > > >

3 3

[alsa-devel] [PATCH 0/8] ASoC updates
by Mark Brown 08 Oct '08

08 Oct '08

The following changes since commit 691ac84a77b1aebb7e09b4924b4b9c07a0f4f66a: Takashi Iwai (1): Merge branch 'asoc-fixes' into topic/asoc are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai This is all 2.6.28ish material. Arun KS (1): ASoC: Add custom SOC_SINGLE_TLV for tlv320aic23 codec Jarkko Nikula (2): ASoC: tlv320aic3x: Use uniform tlv320aic naming ASoC: Remove unused AUDIO_NAME define from codec drivers Mark Brown (4): ASoC: Add WM8753 SPI support ASoC: Add WM8510 SPI support ASoC: Implement WM8510 bias level control ASoC: Make WM8510 microphone input a DAPM mixer Richard Zhao (1): ALSA: ASoC: add new param mux to dapm_mux_update_power sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320aic23.c | 54 ++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.c | 5 +- sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8510.c | 109 ++++++++++++++++++++++++++++++++++----- sound/soc/codecs/wm8510.h | 1 + sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 72 +++++++++++++++++++++++++- sound/soc/codecs/wm8753.h | 1 + sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/soc-dapm.c | 8 ++-- 15 files changed, 221 insertions(+), 37 deletions(-)

2 9

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: dynamic detection of MIDI interfaces in uaxx-quirk
by pedro.lopez.cabanillas＠gmail.com 08 Oct '08

08 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> The MIDI interfaces have to be detected dynamically for Edirol devices ua-700, ua-25 and ua4-fx. This patch reverses the wrong changes made by my other patch in uaxx-quirk. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6e70ba4..bbd70d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3023,6 +3023,31 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); + if (altsd->bNumEndpoints == 2) { + static const struct snd_usb_midi_endpoint_info ua700_ep = { + .out_cables = 0x0003, + .in_cables = 0x0003 + }; + static const struct snd_usb_audio_quirk ua700_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &ua700_ep + }; + static const struct snd_usb_midi_endpoint_info uaxx_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk uaxx_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &uaxx_ep + }; + if (chip->usb_id == USB_ID(0x0582, 0x002b)) + return snd_usb_create_midi_interface(chip, iface, + &ua700_quirk); + else + return snd_usb_create_midi_interface(chip, iface, + &uaxx_quirk); + } + if (altsd->bNumEndpoints != 1) return -ENXIO; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 85532cf..69689e7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -870,11 +870,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1216,11 +1212,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1375,11 +1367,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 -- 1.5.2.4

2 1

[alsa-devel] ASoC WM8510 Mic PGA problem
by ext-jukka.hynninen＠vaisala.com 08 Oct '08

08 Oct '08

Hi all, I'm facing trouble with programmable mic gain stage and WM8510 ASoC driver. First of all, the DAPM control "Mic PGA Switch" seems to be operating just the opposite way the codec datasheet suggests. I inverted the control with expected result, PGA now almost works. When I start capturing, the "Mic PGA" control is instantly turned off again. If I manually enable the control while the capture is running, it works just fine. Any ideas? Am I missing some DAPM routings? -Jukka

3 11

Re: [alsa-devel] [PATCH] Sigmatel stac92hd71b7 low HP volume
by Takashi Iwai 08 Oct '08

08 Oct '08

[please don't drop Cc to alsa-devel ML] At Mon, 6 Oct 2008 12:48:43 +0200, Klaas De Craemer wrote: > > Hello, > > I'm sure that the audioquality will degrade somewhat because of the additional amplification, > but before, the HP-out was not usable at all. I had to use an external ampli just to use my > headphones, even after turning all volume sliders to 100%. Then again, when removing the HP, > the volume though the internal speakers was way too loud... > > I'm not sure what you mean with the output pin and headphone control, but I have uploaded the > output of the codegraph tool before and after my modification here: > http://users.vtk.be/~s0159120/codec_original.svg > http://users.vtk.be/~s0159120/codec_final.svg > > Below is the output of alsa-info with the patched module > Klaas Thanks. So, if I understand correctly, when the pin 0x0a is connected directly from DAC0 0x10, the output level is too low even if you set the all volumes maximal. In the alsa-info.sh output, I see the following: > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] which means that you decrease the output level intentionally. Is the HP output still too low even if you set to the higher value? Takashi > > ----- > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.51 > !!################################ > > !!Script ran on: Mon Oct 6 12:42:48 CEST 2008 > > > !!Linux Distribution > !!------------------ > > Debian GNU/Linux lenny/sid \n \l > > > !!Kernel Information > !!------------------ > > Kernel release: 2.6.26-5.slh.6-sidux-686 > Operating System: GNU/Linux > Architecture: i686 > Processor: unknown > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.16 > Library version: > Utilities version: 1.0.16 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xf6fdc000 irq 21 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > > > !!Advanced information - PCI Vendor/Device/Susbsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:293e (rev 03) > Subsystem: 1028:024f > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-pcsp: index=-2 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : 0 > id : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > single_cmd : N > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: IDT 92HD71B7X > Address: 0 > Vendor Id: 0x111d76b2 > Subsystem Id: 0x1028024f > Revision Id: 0x100302 > No Modem Function Group found > Default PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 > GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 > Node 0x0a [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x081c: OUT HP Detect > Pin Default 0x0421101f: [Jack] HP Out at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=30, enabled=1 > Connection: 3 > 0x10 0x11 0x17* > Node 0x0b [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x04a11221: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0c [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x0e [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x23a1902e: [Jack] Mic at Sep Left > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0xe > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x23014250: [Jack] Line Out at Sep Left > Conn = 1/8, Color = Green > DefAssociation = 0x5, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x3d 0x3d] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x12 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x1c > Processing caps: benign=0, ncoeff=0 > Node 0x13 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x1d > Processing caps: benign=0, ncoeff=0 > Node 0x14 [Pin Complex] wcaps 0x400100: Mono > Pincap 0x0810: OUT > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x16 > Node 0x15 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x10* 0x11 0x17 > Node 0x16 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x15 > Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x1f 0x1f] [0x1f 0x1f] [0x97 0x97] [0x97 0x97] [0x97 0x97] > Connection: 5 > 0x10 0x11 0x27 0x1a 0x1b > Node 0x18 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x90a000f0: [Fixed] Mic at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x19 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x01 0x01] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x00 0x00] > Connection: 4 > 0x1a* 0x17 0x18 0x19 > Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x80 0x80] > Connection: 4 > 0x1b* 0x17 0x18 0x19 > Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x24 > Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital > Pincap 0x0810010: OUT EAPD > EAPD 0x0: > Pin Default 0x40f000f6: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x6 > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Connection: 2 > 0x24* 0x25 > Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x40f000f7: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x7 > Pin-ctls: 0x00: > Connection: 1 > 0x25 > Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x22 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x24 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x21* 0x1c 0x1d > Node 0x25 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x22* 0x1c 0x1d > Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > Node 0x27 [Pin Complex] wcaps 0x400000: Mono > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Node 0x28 [Volume Knob Widget] wcaps 0x600000: Mono > Volume-Knob: delta=1, steps=127, direct=1, val=127 > Connection: 2 > 0x10* 0x11 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw---- 1 root audio 116, 7 okt 6 12:40 /dev/snd/controlC0 > crw-rw---- 1 root audio 116, 6 okt 6 12:40 /dev/snd/hwC0D0 > crw-rw---- 1 root audio 116, 5 okt 6 12:40 /dev/snd/pcmC0D0c > crw-rw---- 1 root audio 116, 4 okt 6 12:41 /dev/snd/pcmC0D0p > crw-rw---- 1 root audio 116, 3 okt 6 11:41 /dev/snd/seq > crw-rw---- 1 root audio 116, 2 okt 6 11:41 /dev/snd/timer > > > !!Aplay/Arecord output > !!------------ > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 2/2 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xf6fdc000 irq 21' > Mixer name : 'IDT 92HD71B7X' > Components : 'HDA:111d76b2' > Controls : 19 > Simple ctrls : 14 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 127 > Mono: Playback 94 [74%] [-24.75dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Front',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [on] > Front Right: Capture 0 [0%] [0.00dB] [on] > Simple mixer control 'Capture Mux',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 1 [33%] [10.00dB] > Front Right: 1 [33%] [10.00dB] > Simple mixer control 'Capture Mux',1 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'Capture',1 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [off] > Front Right: Capture 0 [0%] [0.00dB] [off] > Simple mixer control 'Analog Loopback',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Analog Loopback',1 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Digital Input Source',0 > Capabilities: enum > Items: 'Analog Inputs' 'Digital Mic 1' > Item0: 'Analog Inputs' > Simple mixer control 'Digital Mic 1',0 > Capabilities: cvolume > Capture channels: Front Left - Front Right > Limits: Capture 0 - 3 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > Simple mixer control 'Input Source',1 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.2 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.3 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > value false > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > index 1 > value false > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > value.0 0 > value.1 0 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > index 1 > value.0 0 > value.1 0 > } > control.9 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.10 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > index 1 > value.0 0 > value.1 0 > } > control.11 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Front Playback Volume' > value.0 127 > value.1 127 > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.13 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Headphone Playback Volume' > value.0 127 > value.1 127 > } > control.14 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.15 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Digital Mic 1 Capture Volume' > value.0 0 > value.1 0 > } > control.16 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 'Analog Inputs' > comment.item.1 'Digital Mic 1' > iface MIXER > name 'Digital Input Source' > value 'Analog Inputs' > } > control.17 { > comment.access 'read write' > comment.type INTEGER > comment.count 1 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Master Playback Volume' > value 94 > } > control.18 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Master Playback Switch' > value true > } > control.19 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > comment.dbmin -5100 > comment.dbmax 0 > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > snd_hda_intel > snd_hwdep > nls_cp437 > vfat > fat > loop > isofs > udf > nvidia > ppdev > lp > parport > acpi_cpufreq > cpufreq_performance > cpufreq_stats > cpufreq_ondemand > cpufreq_powersave > cpufreq_conservative > freq_table > ipv6 > af_packet > hfsplus > nls_utf8 > ntfs > nls_base > fuse > snd_pcm_oss > snd_pcm > snd_page_alloc > snd_mixer_oss > pcmcia > firmware_class > snd_seq_dummy > rtc_cmos > snd_seq_oss > rtc_core > rtc_lib > dcdbas > snd_seq_midi_event > snd_seq > snd_timer > snd_seq_device > serio_raw > psmouse > i2c_i801 > snd > pcspkr > evdev > i2c_core > yenta_socket > rsrc_nonstatic > pcmcia_core > soundcore > video > output > wmi > ac > button > battery > intel_agp > ext3 > jbd > mbcache > dm_mirror > dm_log > dm_snapshot > dm_mod > firewire_sbp2 > sg > sr_mod > cdrom > sd_mod > usbhid > hid > ff_memless > ahci > libata > scsi_mod > dock > ricoh_mmc > sdhci > mmc_core > firewire_ohci > firewire_core > crc_itu_t > ehci_hcd > uhci_hcd > usbcore > e1000e > thermal > processor > fan > > > On Monday 06 October 2008 11:44:33 you wrote: > > At Sun, 5 Oct 2008 22:10:46 +0200, > > > > Klaas De Craemer wrote: > > > Hello, > > > > > > A few days ago, I posted a patch to the linux kernel list, but afterwards > > > I realised that it is more appropriate for the alsa list. So I repost it > > > here. > > > > > > Below is my patch for an issue on Dell E6500 notebooks with the Sigmatel > > > stac92hd71b7 codec. In the default snd-hda-intel configuration, the sound > > > going to the headphone jack goes comes directly from the DAC, resulting > > > in very low volume output. The patch changes the codec's widget > > > connections so that the internal mixer is put in between and sets its > > > input amplifiers to the maximum value. The output to the internal speaker > > > port has been left untouched. > > > I have tested this and it gives a much better output. > > > > Thanks for the patch. > > > > I wonder whether this really improves the quality. > > Basically what you does is to set the volume of the input mixer over > > 0dB, and this is usually bad when really high signals come in. > > > > Just to be sure, is the output pin set to the headphone pin control? > > You can give alsa-info.sh output (run with --no-upload option) for > > further analysis. The script is found in > > http://www.alsa-project.org/alsa-info.sh > > > > > > thanks, > > > > Takashi > >

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