January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Staticise twl6040_hs_jack_report()
by Mark Brown 21 Jan '11

21 Jan '11

It's an internal function so shouldn't be exported (as sparse points out). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4bbf1b1..482fcdb 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -724,8 +724,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } -void twl6040_hs_jack_report(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, int report) +static void twl6040_hs_jack_report(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int status; -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: soc-cache: Add trace event for snd_soc_cache_sync()
by Dimitris Papastamos 21 Jan '11

21 Jan '11

This patch makes it easy to see when the syncing process begins and ends. You can also enable the snd_soc_reg_write tracepoint to see which registers are being synced. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/trace/events/asoc.h | 25 +++++++++++++++++++++++++ sound/soc/soc-cache.c | 10 ++++++++++ 2 files changed, 35 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 186e84d..ce993a3 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -229,6 +229,31 @@ TRACE_EVENT(snd_soc_jack_notify, TP_printk("jack=%s %x", __get_str(name), (int)__entry->val) ); +TRACE_EVENT(snd_soc_cache_sync, + + TP_PROTO(struct snd_soc_codec *codec, const char *type, + const char *status), + + TP_ARGS(codec, type, status), + + TP_STRUCT__entry( + __string( name, codec->name ) + __string( status, status ) + __string( type, type ) + __field( int, id ) + ), + + TP_fast_assign( + __assign_str(name, codec->name); + __assign_str(status, status); + __assign_str(type, type); + __entry->id = codec->id + ), + + TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), + (int)__entry->id, __get_str(type), __get_str(status)) +); + #endif /* _TRACE_ASOC_H */ /* This part must be outside protection */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f834839..db66dc4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include <linux/bitmap.h> #include <linux/rbtree.h> +#include <trace/events/asoc.h> + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -1601,18 +1603,26 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); int snd_soc_cache_sync(struct snd_soc_codec *codec) { int ret; + const char *name; if (!codec->cache_sync) { return 0; } + if (codec->cache_ops->name) + name = codec->cache_ops->name; + else + name = "unknown"; + if (codec->cache_ops && codec->cache_ops->sync) { if (codec->cache_ops->name) dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); ret = codec->cache_ops->sync(codec); if (!ret) codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); return ret; } -- 1.7.3.5

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[alsa-devel] [PATCH] ASoC: soc-cache: Add trace event for snd_soc_cache_sync()
by Dimitris Papastamos 21 Jan '11

21 Jan '11

This patch makes it easier to see which of the register writes are part of the cache syncing functionality. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/trace/events/asoc.h | 9 +++++++++ sound/soc/soc-cache.c | 5 +++++ 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 186e84d..7efed81 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -59,6 +59,15 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read, ); +DEFINE_EVENT(snd_soc_reg, snd_soc_cache_sync, + + TP_PROTO(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val), + + TP_ARGS(codec, reg, val) + +); + DECLARE_EVENT_CLASS(snd_soc_card, TP_PROTO(struct snd_soc_card *card, int val), diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f834839..accd96e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include <linux/bitmap.h> #include <linux/rbtree.h> +#include <trace/events/asoc.h> + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -895,6 +897,7 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) codec->cache_bypass = 0; if (ret) return ret; + trace_snd_soc_cache_sync(codec, rbnode->reg, val); dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", rbnode->reg, val); } @@ -1155,6 +1158,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) codec->cache_bypass = 0; if (ret) return ret; + trace_snd_soc_cache_sync(codec, i, val); dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", i, val); } @@ -1415,6 +1419,7 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_write(codec, i, val); if (ret) return ret; + trace_snd_soc_cache_sync(codec, i, val); dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", i, val); } -- 1.7.3.5

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Re: [alsa-devel] [QUERY] [AUDIO][OMAP3 EVM] arecord not working on OMAP3 EVM
by Peter Ujfalusi 21 Jan '11

21 Jan '11

On 01/20/11 17:26, ext Koyamangalath, Abhilash wrote: > Hi all > While testing audio capture (using arecord e,g.),I'm getting only noise in the (zeroed out) output buffers on the OMAP3 EVM. > I have enabled AUXL and AUXR mic inputs through alsamixer, which according to the schematics are connected to the "Audio In" > connector on the board, even though the TWL4030 reg list does not allow me any ways ( or so I think) to specify mic-bias for > these aux inputs. Well, it is auxiliary input (line in). You do not need bias for it. I do not have access to OMAP3 EVM documentation, but Beagle also have AUXL/R as input. I guess you need to use correct cable, and correct source for the input (stereo jack I suppose). > With AUXL and AUXR muted - I get only silence, which seems to make sense. > Is there any basic startup setting I'm missing, maybe specific to OMAP3 EVM? Beagle? > Has anyone seen anything like this before, and could provide pointers? My bet goes for the wrong cable/accessory pair. -- Péter -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 1/2] ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx
by David Henningsson 21 Jan '11

21 Jan '11

Four very similar procedures - one for each model - now refactored into one. This isn't all duplicated code, but a step in the right direction. (And given the next patch, that four procedures would have been five if it weren't for this one...) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH 2/2] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
by David Henningsson 21 Jan '11

21 Jan '11

BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 24 +++++++++++++++++++++++- 2 files changed, 24 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 16ae430..0caf77e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5066 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60 + asus Asus K52JU, Lenovo G560 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7cd59b9..19f0daf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -127,6 +127,7 @@ struct conexant_spec { unsigned int ideapad:1; unsigned int thinkpad:1; unsigned int hp_laptop:1; + unsigned int asus:1; unsigned int ext_mic_present; unsigned int recording; @@ -2312,6 +2313,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_asus_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 0); +} + + /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_hp_laptop_automic(struct hda_codec *codec) { @@ -2400,6 +2414,8 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); + else if (spec->asus) + cxt5066_asus_automic(codec); } /* unsolicited event for jack sensing */ @@ -3045,6 +3061,7 @@ enum { CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3056,6 +3073,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", }; @@ -3068,6 +3086,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3077,6 +3096,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3132,13 +3152,15 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_ASUS: case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; - spec->hp_laptop = 1; + spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; + spec->asus = board_config == CXT5066_ASUS; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ -- 1.7.1 --------------050308060108010003060101--

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[PATCH 1/2] ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx
by David Henningsson 21 Jan '11

21 Jan '11

Four very similar procedures - one for each model - now refactored into one. This isn't all duplicated code, but a step in the right direction. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 81 +++++++++++---------------------------- 1 files changed, 23 insertions(+), 58 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9bb030a..7cd59b9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2387,79 +2387,53 @@ static void cxt5066_hp_automute(struct hda_codec *codec) cxt5066_update_speaker(codec); } -/* unsolicited event for jack sensing */ -static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) +/* Dispatch the right mic autoswitch function */ +static void cxt5066_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - /* ignore mic events in DC mode; we're always using the jack */ - if (!spec->dc_enable) - cxt5066_olpc_automic(codec); - break; - } -} -/* unsolicited event for jack sensing */ -static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + if (spec->dell_vostro) cxt5066_vostro_automic(codec); - break; - } -} - -/* unsolicited event for jack sensing */ -static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + else if (spec->ideapad) cxt5066_ideapad_automic(codec); - break; - } + else if (spec->thinkpad) + cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } /* unsolicited event for jack sensing */ -static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + struct conexant_spec *spec = codec->spec; + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_hp_laptop_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } /* unsolicited event for jack sensing */ -static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_thinkpad_automic(codec); + cxt5066_automic(codec); break; } } + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -3039,20 +3013,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: init\n"); conexant_init(codec); if (codec->patch_ops.unsol_event) { cxt5066_hp_automute(codec); - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); + cxt5066_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3169,7 +3134,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; @@ -3207,7 +3172,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_DELL_VOSTRO: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_vostro_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; @@ -3224,7 +3189,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_IDEAPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_ideapad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_ideapad; @@ -3240,7 +3205,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_THINKPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_thinkpad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; -- 1.7.1 --------------090909050907040908000909--

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Re: [alsa-devel] [PATCH - [hdspm] 2/3] Enable RME RayDAT and AIO
by Takashi Iwai 21 Jan '11

21 Jan '11

At Tue, 18 Jan 2011 15:12:18 +0100, Adrian Knoth wrote: > > @@ -29,12 +42,13 @@ > #include <linux/moduleparam.h> > #include <linux/slab.h> > #include <linux/pci.h> > -#include <linux/math64.h> No math64 stuff is necessary? > +#include <linux/sysfs.h> It's not implemented yet, so don't include it. > -static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { > +static struct pci_device_id snd_hdspm_ids[] __devinitdata = { Why open-coded again? > @@ -796,6 +1009,8 @@ static void hdspm_silence_playback(struct hdspm *hdspm) > int n = hdspm->period_bytes; > void *buf = hdspm->playback_buffer; > > + snd_printk(KERN_INFO "hdspm_silence_playback\n"); Don't put such a verbose debug message in frequently called callbacks. > static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) ... > n = div_u64(n, rate); > /* n should be less than 2^32 for being written to FREQ register */ > - snd_BUG_ON(n >> 32); > + snd_BUG_ON((n >> 32) == 0); This check is consistent with the comment above... The rest is too long to review. Would be nice if you can split the whole patch in logical orders... thanks, Takashi

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[alsa-devel] Question about ASOC codec drivers
by Neil Jones 20 Jan '11

20 Jan '11

Hi, Im trying to write an ASOC Codec driver for a new codec and im having problems setting up the widgets and interconnects. The input structure is similar to the wm8731 so i have based my codec on this (my widget code is included at the bottom) problem is I get this error on startup: Failed to add route Line Input->Input Mux Add then when I run alsa-mixer I get a memfault in the strcpy in snd_soc_info_enum_double() it appears to be indexing a string 1 past the end, ie uinfo->value.enumerated.item = 3 but e->texts is pointing to the tansen_input_select[] array of strings which only has 3 entries, so it blows up (e->max = 7??). also shouldn't the strcpy by a strncpy as we are copying to a fixed 64byte buffer ? thanks for anyhelp. Neil my widget code: static const struct snd_kcontrol_new tansen_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TANSEN_TM_OP_TOP7, TANSEN_TM_OP_TOP8, 0, 0xFF, 0 ), SOC_DOUBLE("Line/IPOD-In Gain", TANSEN_TM_IP_TOP5, 5, 2, 0x7, 0), SOC_DOUBLE("Mic-In Gain", TANSEN_TM_IP_TOP4, 0, 4, 0xF, 0), }; static const char *tansen_input_select[] = { "Mic In", "Line In", "IPOD In", }; static const struct soc_enum tansen_ip_enum = SOC_ENUM_SINGLE(TANSEN_TM_IP_TOP2, 4, 0x7, tansen_input_select); /* Input mux */ static const struct snd_kcontrol_new tansen_input_mux_controls = SOC_DAPM_ENUM("Input Select", tansen_ip_enum); static const struct snd_soc_dapm_widget tansen_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &tansen_input_mux_controls), SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", TANSEN_TM_MICBIAS_1, 4, 1), SND_SOC_DAPM_INPUT("MICIN"), SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("LLINEIN"), SND_SOC_DAPM_INPUT("RIPODIN"), SND_SOC_DAPM_INPUT("LIPODIN") }; static const struct snd_soc_dapm_route intercon[] = { /* outputs */ {"RHPOUT", NULL, "DAC"}, {"LHPOUT", NULL, "DAC"}, /* input mux */ {"Input Mux", "Line In", "Line Input"}, {"Input Mux", "Mic In", "Mic bias"}, {"Input Mux", "Ipod In", "Ipod Input"}, {"ADC", NULL, "PGA"}, {"PGA", NULL, "Input Mux"}, /* inputs */ {"Line Input", NULL, "LLINEIN"}, {"Line Input", NULL, "RLINEIN"}, {"Ipod Input", NULL, "LIPODIN"}, {"Ipod Input", NULL, "RIPODIN"}, {"Mic bias", NULL, "MICIN"}, };

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[alsa-devel] Automatically disabling internal speakers when headphones are plugged in
by Stephen Warren 20 Jan '11

20 Jan '11

I'm working on jack plug detect for Tegra Harmony, and have a question about automatically disabling speakers when a headphone jack is plugged in. For test purposes, I implemented this as follows: static struct snd_soc_jack_pin harmony_hp_jack_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, }, { .pin = "Int Spk", .mask = SND_JACK_HEADPHONE, .invert = 1, }, }; ... following the example of: omap/ams-delta.c samsung/rx1950_uda1380.c samsung/h1940_uda1380.c However, I wonder if that's the correct approach; should the plug detect logic only control the headphone output (i.e. only include the first array entry in jack_pins[] above), and leave it up to user-space whether to disable the speaker in this case, using a custom Speaker Function control created by the machine driver, as I see many other drivers expose? Do the jack reports make it up to user-space so that could be implemented? Thanks. -- nvpublic

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