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[alsa-devel] Very faint sound on Dell Latitude E6500
by Laurent Pinchart 30 Jan '09

30 Jan '09

Hi everybody, I've received a brand new Dell Latitude E6500 and found out that the sound card wasn't exactly well supported by ALSA. Sound is barely audible (so faint that I'm not even sure if the proper sound is output or if I just get garbage), with some loud noise here and there that seem to correspond to the saturated audio data parts. The sound card is an HDA-Intel with an IDT 92HD71B7X codec. Hardware information can be found at http://www.alsa-project.org/db/?f=30cac30e5b1ddd9c842481a830c250cc3e593b0d. A bug report including the Windows driver .ini file is available at https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i…. When loading the snd_hda_intel driver compiled with verbose debug mode, the following messages are printed to the kernel log. HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA sound/pci/hda/hda_intel.c:2209: chipset global capabilities = 0x4401 ALSA sound/pci/hda/hda_intel.c:775: codec_mask = 0x5 ALSA sound/pci/hda/hda_codec.c:2367: hda_codec: model 'dell-m4-1' is selected for config 1028:24f (unknown Dell) ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0a pin config 0421101f ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0b pin config 04a11221 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0c pin config 40f000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0d pin config 90170110 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0e pin config 23a1902e ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0f pin config 23014250 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 14 pin config 40f000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 18 pin config 90a000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 19 pin config 40f000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 1e pin config 4f0000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 1f pin config 4f0000f0 ALSA sound/pci/hda/hda_codec.c:3055: autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3059: speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3063: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3064: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3072: inputs: mic=0xb, fmic=0xe, line=0x0, fline=0x0, cd=0x0, aux=0x0 ALSA sound/pci/hda/patch_sigmatel.c:2693: stac92xx_add_dyn_out_pins: total dac count=2 ALSA sound/pci/hda/hda_generic.c:679: hda_generic: no proper input path found ALSA sound/pci/hda/hda_generic.c:284: Skip Digital OUT node 2 ALSA sound/pci/hda/hda_generic.c:427: hda_generic: no proper output path found ALSA sound/pci/hda/hda_generic.c:1022: hda_generic: no PCM found ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Surround Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Center Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave LFE Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Side Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave External Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker2 Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Surround Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Center Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave LFE Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Side Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave External Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker2 Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave IEC958 Playback Switch, skipped I tried forcing the model to dell-m4-1, dell-m4-2, dell-m4-3 and ref without any luck. I also tried upgrading to alsa-driver 1.0.19 and results were identical. Could anyone help me ? I can patch and recompile the driver is required. Best regards, Laurent Pinchart

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[alsa-devel] kerneldoc for sound drivers?
by Timur Tabi 30 Jan '09

30 Jan '09

I'm fixing the comments in my driver (sound/soc/codecs/cs4270.c) to conform to kerneldoc standards. However, when I build "make htmldocs", my comments aren't processed. I presume the make rules don't know to look into the sound/soc/codecs directory. How do I fix this? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Usb driver debug
by nick83ola 30 Jan '09

30 Jan '09

anyone can give me information about how i can debug the driver for my usb edirol ua-20 ?(that not works in advanced mode...sigh sob) i have downloaded the kernel source and locked in sound/usb any examples about how to debug/develop usb linux drivers??? Please I need it!!!!! Nicola Lunghi -- www.nicolalunghi.it http://www.cerchiopisa3.netsons.org P.S. Le informazioni trasmesse attraverso la presente comunicazione sono di sclusiva spettanza dell'effettivo destinatario. Nel caso in cui le stesse raggiungessero, per qualunque motivo, soggetti non interessati, questi ultimi vorranno darne immediata notizia al mittente. In ogni caso, eventuali soggetti diversi dai legittimi destinatari della presente comunicazione e dei dati contenuti negli allegati, possono essere sanzionati ai sensi del T.U. sul trattamento dei dati personali d.lgs. 196/2003, sia ai sensi dell'art. 616 del Codice Penale che disciplina la violazione del segreto sulla corrispondenza.

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[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 30 Jan '09

30 Jan '09

Hello, I walked through the code for snd-usb-audio and have a fair idea about the structure of the driver. I also went through the "writing-an-alsa-driver" article on http://www.alsa-project.org Here is the stuff I've done so far http://mndar.phpnet.us/usbxfi/ .The link contains the "lsusb -v" output for the Creative USB X-Fi 5.1 . The xfi-0.0.0.0.tar.bz2 has the basic structure of an alsa driver (copied from the caiaq usb driver). It presently just loads the module when the card is connected. Output in /var/log/messages ------------------------------------------- Jan 30 10:46:29 mandar Found Creative USB X-Fi 5.1!! Jan 30 10:46:29 mandar 0 041e:3042 Jan 30 10:46:29 mandar Found Creative USB X-Fi 5.1!! Jan 30 10:46:29 mandar 0 041e:3042 Jan 30 10:46:29 mandar Found Creative USB X-Fi 5.1!! Jan 30 10:46:29 mandar 0 041e:3042 Jan 30 10:46:29 mandar usbcore: registered new interface driver snd-usb-xfi -------------------------------------------------------------------------------- As you can see, the probe is being called 3 times. Can someone please go through the source and let me know if I'm headed in the right direction? I'm a newbie to alsa driver development and would really appreciate any input from you guys. Also, please have a look the creative.txt file in the link. Its the 'lsusb -v' output for the card. How exactly do I interpret and use this information? Regards Mandar Joshi

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[alsa-devel] No digital output on HP Pavilion dv6936us (intel_hda ALC268), also volume low--help me hack?
by Jens Jorgensen 30 Jan '09

30 Jan '09

Hello, I've got sound working on my laptop but there are two problems I'd like to solve. One is the volume is kind of low. Not so low I'm positive something is wrong, but it seems like it isn't right. Second, and more interesting, is this laptop seems to digital audio output that goes over the HDMI connector. Or, I should say, in Windows XP (installing that was another adventure I would have preferred not to have done) The mixer has a control to enable S/PDIF output. I'm a (non-kernel) c++ developer by trade and I've got a local kernel tree from git and should be able to manage pulling the alsa git as well, so if some alsa person could guide me along in hacking the driver to help me try to get this working that would be great. I ran alsa-info.sh on my box, here's the URL: http://www.alsa-project.org/db/?f=1b5a45d1336cee176dbd99e725a56f7ad4e20b42 -- Jens B. Jorgensen jbj1(a)ultraemail.net

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[alsa-devel] Can DMA address, periods, buffer size, etc change after hw_params is called?
by Hans-Christian Egtvedt 29 Jan '09

29 Jan '09

Hi, I wonder if I must take care handling changes in DMA address, number of periods and period size in the prepare callback? Or are these locked after the first time prepare is called? The reason I ask is because I'm developing a new ALSA driver and I prepare a cyclic DMA transfer in the first prepare callback. Additional calls to prepare will not do anything DMA stuff. The cyclic DMA transfer is then released in the close callback. Should I after the open and hw_params calls be ready to handle any change at all regarding DMA buffer address, periods and/or number of periods? I hope I made sense in this email (-: -- Best regards, Hans-Christian Egtvedt

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Analog bypass implementation - try2
by Peter Ujfalusi 29 Jan '09

29 Jan '09

Hello, Second try for the analog bypass implementation. Since the first try most of the things has been turned around - I tried to follow Mark's advice, I hope I managed well enough: [1] New power, codec state handling The codec will be powered at all time, except if the codec's bias level has been set to SND_SOC_BIAS_OFF via the twl4030_set_bias_level call. [2] Keeping the power consumption on the lowest level When the codec is in SND_SOC_BIAS_ON state (playback/recording), there is nothing can be done, the amplifiers are powered, PLL is enabled. When the codec enters to SND_SOC_BIAS_STANDBY state, two thing can happen: - If none of the bypass switches are enabled, than the gain controls for the outputs are forced to power-down mode and the PLL is disabled - If one of the bypass switches are enabled, than the gain controls will be kept in their user configured state and the PLL stays enabled. When the codec is in SND_SOC_BIAS_STANDBY (gains and PLL are disabled), change in the bypass config will trigger an event and if it is needed the gains and PLL will be enabled. Switching all bypass off will switch the gains and PLL off. With the this implementation, there is no visible difference in power consumption compared to the previous, codec off way. So when the codec is in SND_SOC_BIAS_STANDBY mode and all of the bypasses are disabled, the additional power consumption is minimal, if any. --- Peter Ujfalusi (1): ASoC: TWL4030: Add analog loopback support sound/soc/codecs/twl4030.c | 212 +++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl4030.h | 15 +++ 2 files changed, 214 insertions(+), 13 deletions(-)

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[alsa-devel] [PATCH] ALSA: hda - make alc882_auto_init_input_src aware of selectors
by Herton Ronaldo Krzesinski 29 Jan '09

29 Jan '09

In the case of having a selector instead of mixer while initing input sources, the case that happens with ALC889, we must select instead of muting input. Problem was found while testing with hda-emu. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56adbb8..31a6d40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6924,18 +6924,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) static void alc882_auto_init_input_src(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; int c; for (c = 0; c < spec->num_adc_nids; c++) { hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; int conns, mute, idx, item; conns = snd_hda_get_connections(codec, nid, conn_list, ARRAY_SIZE(conn_list)); if (conns < 0) continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; for (idx = 0; idx < conns; idx++) { /* if the current connection is the selected one, * unmute it as default - otherwise mute it @@ -6948,8 +6951,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) break; } } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute); + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } -- 1.6.1

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[alsa-devel] CS4232 Playback bug on Tecra 500
by Stanley Pinchak 29 Jan '09

29 Jan '09

I have been running into the problem briefly mentioned at the wiki entry http://alsa.opensrc.org/index.php/Toshiba_Tecra_500CDT. Playback using the CS4232 driver in native or oss emulation mode fails randomly. Once the failure occurs, subsequent attempts to playback stop shortly after opening the device and tickling the speakers (perhaps the buffer gets filled once). Under heavy network load, the initial failure will occur rather soon, within 15 minutes or so of playback. When I am playing songs from the hard drive, the failure takes a very long time, and when I use oss emulation and play from the harddrive, I have been unable to replicate the bug. The CPU does have the F00F bug if that might make some difference. I have attached a backtrace resulting from the attempts to use aplay after the failure has occurred. I may be able to attempt to capture a backtrace at the initial failure if this would be helpful. There are no obvious error messages in my logfiles or printed to the console when the bug occurs. Please let me know how I can assist in squashing this bug. I attempted to use the alsa-info script, but it appears that the database where it uploads the results is broken at the moment. I have pastebinned the results at http://pastebin.ca/1315459 Thanks Stanley Pinchak

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[alsa-devel] alsa-utils compile issues
by Will Wagner 29 Jan '09

29 Jan '09

Hi All, I have configured alsa-lib 1.0.19 to remove the components I do not want for my embedded arm target. I do this with the configure options: --disable-docs \ --disable-aload \ --disable-rawmidi \ --disable-hwdep \ --disable-seq \ --disable-alisp \ --disable-old-symbols \ --disable-python \ --with-alsa-devdir=/dev \ --with-pcm-plugins="copy linear adpcm rate plug dmix softvol" \ --with-ctl-plugins="" \ --with-versioned="no" \ This builds a much reduced in size alsa-lib However when I then try to build alsa-utils this fails because amidi and seq will not compile with the reduced alsa-lib It seems to be there should be configure options in alsa-utils to disable more of its components than just alsamixer and alsatest. I have worked around this by just removing amidi and seq from the SUBDIRS variable in Makefile.in but perhaps there is a better way of doing this? Thanks, Will -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Development Manager Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------

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