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

[alsa-devel] SB X-Fi driver status update
by Takashi Iwai 17 Oct '08

17 Oct '08

Hi, now I'm stabilizing the driver a bit. I hope it won't hang up any more (so often), and make you happy testing. ** DOWNLOAD ** The snd-sbxfi driver is included in my sound-unstable git tree, and alsa-driver-unstable snapshot tarball. git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-unstable-2.6.git ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… The git commits can be browsed via http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable-2.6.git;a=su… The snapshot tarball includes HEAD and alsa-kernel/HEAD files. These contain alsa-driver and sound git commit logs. Check these files and compare with the above gitweb page whether you have the latest version. ** BUILD ** If you want to build the sbxfi driver on your 2.6.27 git tree, at best, do the following: - pull sound-2.6.git tree devel branch % git pull \ git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git devel - pull sound-unstable-2.6.git tree topic/sbxfi branch % git pull \ git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-unstable-2.6.git\ topic/sbxfi In that way, you can update only the sound and keep the rest clean. (The master branches of sound and sound-unstable git trees are for the latest upsteram, i.e. 2.6.28-pre git tree.) In other versions, it'd be better to use alsa-driver-unstable snapshot tarball to build external modules. ** UPDATED NOTES ** Below are some updates: - As default, the driver works only as a constant sample rate, either 48000 or 96000 (default). The rate can be changed via the module option base_rate. For enabling the continuous rate support (it was on before), define XXX_CONT_RATE in sbxfi.c. - The debug module option is 1 as default, i.e. showing some debug messages but not too often. You can set debug=0 to suppress, or debug=2 or 3 to see more. If you have problems (especially regarding with the sound quality problem), run with debug=2. - Please check whether the default (base_rate=96000) works as is for different rates, formats, channels. Then try base_rate=48000. - If you are brave, try recording, too :) It's no full-duplex, so you can't play and record at the same time. Define XXX_FULL_DUPLEX in sbxfi.c to enable the (possible) full-duplex. - If playback works well, try to define XXX_USE_SG in sbxfi.c. This will make the driver using SG buffer instead of continuous pages. - Also, test SBXFI_DMA_MASK being DMA_32BIT_MASK, especially if you have RAM more than 2GB. - You can try dmix as well. E.g. simply run like % aplay -Dplug:dmix:0 foo.wav thanks, Takashi

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Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Bjoern Olausson 17 Oct '08

17 Oct '08

I found a site with a large range of files: http://bellatrix.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html aplay plays all files it is capable to decode. No crackel, no noise, just fine. glass.wav Signed 16 bit Little Endian, Rate 8000 Hz, Mono M1F1-int16-AFsp.wav Signed 16 bit Little Endian, Rate 8000 Hz, Stereo M1F1-int24-AFsp.wav Signed 24 bit Little Endian in 3bytes, Rate 8000 Hz, Stereo M1F1-int32-AFsp.wav Signed 32 bit Little Endian, Rate 8000 Hz, Stereo M1F1-uint8-AFsp.wav Unsigned 8 bit, Rate 8000 Hz, Stereo stereol.wav Signed 16 bit Little Endian, Rate 22050 Hz, Stereo Utopia_Critical_Stop.wav Signed 16 bit Little Endian, Rate 22050 Hz, Mono Thought when I try to play the first one with mplayer ---> FREEZZZZZ Spiele glass.wav. Audio file-Dateiformat erkannt! ========================================================================== Öffne Audiodecoder: [pcm] Uncompressed PCM audio decoder AUDIO: 8000 Hz, 1 ch, s16le, 128.0 kbit/100.00% (ratio: 16000->16000) Ausgewählter Audiocodec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Amarok with backend ALSA plays all files it can decode without problems. All Signed 16 bit Little Endian, Rate 8000 to 96000 Hz, Stereo do work (with the same oszilating noise described in my previous post with the mono files [I just joned two to get stereo])

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[alsa-devel] HD-audio reconfigure branch merged to master
by Takashi Iwai 16 Oct '08

16 Oct '08

Hi, FYI, I merged the long-pending hda-reconfig topic branch to master and devel branches of sound.git tree for 2.6.29 kernel (not 28). This allows you to reconfigure the HD-audio setup dynamically by triggering via sysfs entries. I'll give more details later... Anyway, let me know if you encounter any build problems or Oops by these changes. thanks, Takashi

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[alsa-devel] Salsa-lib : _snd_pcm_hw_param_test is not defined
by Matthieu CASTET 16 Oct '08

16 Oct '08

Hi, it seems there is a missing implementation of _snd_pcm_hw_param_test function in src/pcm_params.c . It is used in pcm_macros.h by snd_pcm_hw_params_test_xxx functions. Thanks Matthieu

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[alsa-devel] [PATCH 1/2] ASoC: Convert playpaq_wm8510 to bulk route registration API
by Mark Brown 16 Oct '08

16 Oct '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/at32/playpaq_wm8510.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 98a2d58..b1966e4 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to SPKOUT */ {"Ext Spk", NULL, "SPKOUTP"}, {"Ext Spk", NULL, "SPKOUTN"}, @@ -312,9 +312,6 @@ static const char *intercon[][3] = { {"Mic Bias", NULL, "Int Mic"}, {"MICN", NULL, "Mic Bias"}, {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, }; @@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) /* * Setup audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + /* always connected pins */ -- 1.5.6.5

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[alsa-devel] Status of HDMI on Nvidia 10de:0774 device (Asus M3N-H HDMI
by Blacky (alsa-devel) 16 Oct '08

16 Oct '08

Hi all, I am using the above model (MB Asus M3N-H/HDMI) and attempting to use the latest patches for nvhdmi. (By latest, I just checked them from CVS today, and they seem to be very recent.. very bleeding edge) Do they support the 10de:0774 device? The source matches them against 0x10de0004, but I've not been able to identify if it is matching or not. (There are mask values in there which I've not worked out yet). So far, I've failed abysmally at getting HDMI (audio) output on the above mentioned MB. Can anyone tell me if the code is even there to support it? I'm starting to suspect it isn't. I'll follow up with details (aplay -L / -l, asoundconf and other details if there is interest.) Getting the HDMI audio working will be the final piece of the puzzle stopping me from putting the new system into the living room as a working mythtv box. Regards Blacky

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Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Alexey Bazhin 16 Oct '08

16 Oct '08

Hi, I have just compiled driver from git and try to run it. My card is pci: 1102:0005 Multimedia audio controller: Creative Labs SB X-Fi I got following: root@mayonaka:~# modprobe snd-sbxfi root@mayonaka:~# speaker-test -c 2 -r 96000 -t sine speaker-test 1.0.15 Playback device is default Stream parameters are 96000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 96000Hz (requested 96000Hz) [ 262.522739] XFi DAC reset timeout Period size ran[ 262.564029] XFi DAC reset timeout ge from 32 to 16[ 262.620275] XFi DAC reset timeout 384 Using max b[ 262.676529] XFi DAC reset timeout uffer size 16384[ 262.732784] XFi DAC reset timeout Periods = 4 w[ 262.789038] XFi DAC reset timeout as set period_size = 4096 was set buffer_size = 16384 0 - Front Left [ 327.491997] BUG: soft lockup - CPU#0 stuck for 61s! [swapper:0] [ 327.492001] Kernel panic - not syncing: softlockup: hung tasks and i did hear the sine on the left channel. Hope it helps... -- Alexey Bazhin mailto:baz@irc.msk.ru ICQ 125125882

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[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Bjoern Olausson 16 Oct '08

16 Oct '08

>Hi, > >$SUBJECT is now on my sound-unstable git tree: > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound->unstable-2.6.git >together with other experimental patches. > Just wanted to help out with testing stuff. I compiled your snapshot on my gentoo Linux x86_64 Compilation, installation and loading the drivers went smooth. And, yes, there is sound... Here some infos: blub@freax $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ORIG [SB XFi (ORIG)], device 0: SB-XFi [SB-XFi] Subdevices: 1/1 Subdevice #0: subdevice #0 blub@freax $ lspci -v 01:00.0 Multimedia audio controller: Creative Labs SB X-Fi Subsystem: Creative Labs X-Fi Platinum Flags: bus master, medium devsel, latency 64, IRQ 21 I/O ports at 8c00 [size=32] Memory at eb800000 (64-bit, non-prefetchable) [size=2M] Memory at e4000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Kernel driver in use: SB-XFi root@freax $ lsmod | grep snd snd_pcm_oss 40640 0 snd_mixer_oss 17224 1 snd_pcm_oss snd_seq_oss 33168 0 snd_seq_midi_event 9864 1 snd_seq_oss snd_seq 54056 4 snd_seq_oss,snd_seq_midi_event snd_seq_device 10076 2 snd_seq_oss,snd_seq snd_hda_intel 505276 0 snd_hwdep 10320 1 snd_hda_intel snd_sbxfi 23132 1 snd_pcm 81328 3 snd_pcm_oss,snd_hda_intel,snd_sbxfi snd_timer 24160 2 snd_seq,snd_pcm snd 65128 12 snd_pcm_oss,snd_mixer_oss,snd_seq_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_hwdep,snd_sbxfi,snd_pcm,snd_timer soundcore 10208 1 snd snd_page_alloc 11664 3 snd_hda_intel,snd_sbxfi,snd_pcm root@freax $ emerge --info Portage 2.1.4.5 (default-linux/amd64/2007.0/desktop, gcc-4.1.2, glibc-2.6.1-r0, 2.6.27-unpatched x86_64) So now the funny part: I was testing a few samplerates: Used a 1 kHz sine, 0 dBFS, 16 bit mono wave file, 96 kHz sample rate and rsampled it with sox: for i in 88200 50400 50000 48000 47250 44100 44056 32000 22050 11025 8000 ; do sox 0_16_96000.wav -r ${i} 0_16_${i}.wav ; done and played them with aplay: for i in 96000 88200 50400 50000 48000 47250 44100 44056 32000 22050 11025 8000 ; do echo "Now playing 0_16_${i}.wav" ; aplay 0_16_${i}.wav ; echo "DONE" ; done 96000 --> O.K. 88200 --> No Sound 50400 --> No Sound 50000 --> No Sound 48000 --> O.K. 47250 --> No Sound 44100 --> No Sound 44056 --> No Sound 32000 --> No Sound 22050 --> SYSTEM FREEZ Playing WAVE '0_16_22050.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono 11025 --> SYSTEM FREEZ Playing WAVE '0_16_11025.wav' : Signed 16 bit Little Endian, Rate 11025 Hz, Mono 8000 --> No Sound Let me know if I can help out testing some more stuff kind regards Bjoern

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[alsa-devel] [PATCH] kernel docs: fix sound/core/ kernel-doc
by Randy Dunlap 16 Oct '08

16 Oct '08

From: Randy Dunlap <randy.dunlap(a)oracle.com> Add kernel-doc function short descriptions to sound/core functions that are missing this short description. Mostly this involves moving some of the function description onto the @funcname line. Also correct a few variable names and fix other kernel-doc notation. Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 48 ++++++++++++++++++++++++------------------------ sound/core/pcm_native.c | 24 ++++++++---------------- 2 files changed, 32 insertions(+), 40 deletions(-) --- lin2627-g5-kdoc.orig/sound/core/pcm_lib.c +++ lin2627-g5-kdoc/sound/core/pcm_lib.c @@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_r EXPORT_SYMBOL(snd_pcm_hw_rule_add); /** - * snd_pcm_hw_constraint_mask + * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. */ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask) @@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct sn } /** - * snd_pcm_hw_constraint_mask64 + * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the 64bit bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. */ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask) @@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct } /** - * snd_pcm_hw_constraint_integer + * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the integer constraint * @@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); /** - * snd_pcm_hw_constraint_minmax + * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the range * @min: the minimal value @@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct s /** - * snd_pcm_hw_constraint_list + * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the list constraint @@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struc } /** - * snd_pcm_hw_constraint_ratnums + * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint @@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struc } /** - * snd_pcm_hw_constraint_ratdens + * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint @@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct } /** - * snd_pcm_hw_constraint_msbits + * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule * @runtime: PCM runtime instance * @cond: condition bits * @width: sample bits width @@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct s } /** - * snd_pcm_hw_constraint_step + * snd_pcm_hw_constraint_step - add a hw constraint step rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the step constraint @@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct s } /** - * snd_pcm_hw_constraint_pow2 + * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint @@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_p EXPORT_SYMBOL(_snd_pcm_hw_params_any); /** - * snd_pcm_hw_param_value + * snd_pcm_hw_param_value - return @params field @var value * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Return the value for field PAR if it's fixed in configuration space - * defined by PARAMS. Return -EINVAL otherwise + * Return the value for field @var if it's fixed in configuration space + * defined by @params. Return -%EINVAL otherwise. */ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) @@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struc /** - * snd_pcm_hw_param_first + * snd_pcm_hw_param_first - refine config space and return minimum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values > minimum. Reduce configuration space accordingly. * Return the minimum. */ @@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct /** - * snd_pcm_hw_param_last + * snd_pcm_hw_param_last - refine config space and return maximum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values < maximum. Reduce configuration space accordingly. * Return the maximum. */ @@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm EXPORT_SYMBOL(snd_pcm_hw_param_last); /** - * snd_pcm_hw_param_choose + * snd_pcm_hw_param_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * - * Choose one configuration from configuration space defined by PARAMS + * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time --- lin2627-g5-kdoc.orig/sound/core/pcm_native.c +++ lin2627-g5-kdoc/sound/core/pcm_native.c @@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_start + * snd_pcm_start - start all linked streams * @substream: the PCM substream instance - * - * Start all linked streams. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_stop + * snd_pcm_stop - try to stop all running streams in the substream group * @substream: the PCM substream instance * @state: PCM state after stopping the stream * - * Try to stop all running streams in the substream group. - * The state of each stream is changed to the given value after that unconditionally. + * The state of each stream is then changed to the given state unconditionally. */ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) { @@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substrea EXPORT_SYMBOL(snd_pcm_stop); /** - * snd_pcm_drain_done + * snd_pcm_drain_done - stop the DMA only when the given stream is playback * @substream: the PCM substream * - * Stop the DMA only when the given stream is playback. - * The state is changed to SETUP. + * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) @@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_suspend + * snd_pcm_suspend - trigger SUSPEND to all linked streams * @substream: the PCM substream * - * Trigger SUSPEND to all linked streams. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend(struct snd_pcm_substream *substream) @@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_subst EXPORT_SYMBOL(snd_pcm_suspend); /** - * snd_pcm_suspend_all + * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm * @pcm: the PCM instance * - * Trigger SUSPEND to all substreams in the given pcm. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend_all(struct snd_pcm *pcm) @@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_prepare + * snd_pcm_prepare - prepare the PCM substream to be triggerable * @substream: the PCM substream instance * @file: file to refer f_flags - * - * Prepare the PCM substream to be triggerable. */ static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file)

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[alsa-devel] [PATCH] hda: add support for jack detection on IDT/Sigmatel
by Matthew Ranostay 16 Oct '08

16 Oct '08

Added support for jack detection reporting to userspace for IDT/Sigmatel codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/Kconfig b/pci/Kconfig index 7003711..7e40890 100644 --- a/pci/Kconfig +++ b/pci/Kconfig @@ -501,6 +501,7 @@ config SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER + select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for Intel "High Definition Audio" (Azalia) motherboard devices. diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index a2ac720..f6c0b1d 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -30,6 +30,7 @@ #include <linux/pci.h> #include <sound/core.h> #include <sound/asoundef.h> +#include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" @@ -216,6 +217,9 @@ struct sigmatel_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ + /* jack detection */ + struct snd_jack *jack; + /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; unsigned int num_kctl_alloc, num_kctl_used; @@ -3617,7 +3621,7 @@ static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int i; + int i, err; snd_hda_sequence_write(codec, spec->init); @@ -3639,6 +3643,12 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], AC_PINCTL_OUT_EN); stac92xx_auto_init_hp_out(codec); + /* jack detection */ + err = snd_jack_new(codec->bus->card, + "Headphone Jack", + SND_JACK_HEADPHONE, &spec->jack); + if (err < 0) + return err; /* fake event to set up pins */ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); } else { @@ -3796,6 +3806,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } + snd_jack_report(spec->jack, + presence ? SND_JACK_HEADPHONE : 0); if (presence) { /* disable lineouts, enable hp */

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