April 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: tegra: drop Kconfig description for SND_SOC_TEGRA_DAS
by Stephen Warren 05 Apr '12

05 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The DAS, I2S, and SPDIF Kconfig options are intended to be selected by the Kconfig options for ASoC machine drivers. As such, they don't need to be user-visible themselves. Drop the description from the DAS variable to achieve this. I2S and SPDIF already don't have a description. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 23ab00d..16632b1 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -5,7 +5,7 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA_DAS - tristate "Tegra Digital Audio Switch driver" + tristate depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra DAS module. -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: set idle_bias_off=1 for all platform DAPM contexts
by Stephen Warren 05 Apr '12

05 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The ASoC core currently defaults to using STANDBY rather than OFF for idle ASoC platform devices, which causes a permanent pm_runtime_get() on them. This keeps the device active unnecessarily. This can be especially problematic when the ASoC platform device and DAI device are the same device. The distinction between OFF and STANDBY is likely not relevant for ASoC platform drivers, since they aren't analog devices. So, solve this issue by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this turns out to be a problem, this value could be sourced from the snd_soc_platform_driver, similarly to soc_probe_codec(). Note: Prior to this change, this caused a large (10) runtime_active count for the Tegra I2S controller even when not in use, and a leak in that value as streams were started and stopped. This change probably hides a bug. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8a17048..63e6114 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1072,6 +1072,8 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); } + platform->dapm.idle_bias_off = 1; + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { -- 1.7.0.4

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[alsa-devel] Alsa soc UDA1380 - no duplex
by jarek 05 Apr '12

05 Apr '12

Hello! I've received information from lpclinux.com that uda1380 on lpc313x doesn't support duplex. I've got some information that uda1380 on s3c24xx works with duplex, so I think, that it should work also on lpc313x. I'm ready to write some code, but I have no experience with ALSA, so I'll be happy if someone will give me some hints where to start. The other question is why there are two drivers to the same device. best regards Jarek

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[alsa-devel] [PATCH] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 05 Apr '12

05 Apr '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, use a separate mixer control for the inverted channel. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 88 ++++++++++++++++++++++++++++++++++------ 1 files changed, 75 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e6eafb1..c18e3b1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4107,9 +4108,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4119,7 +4120,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4138,7 +4139,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4208,6 +4209,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) return 0; } +/* Returns zero if this is a normal stereo channel, and non-zero if it should + be split in two independent channels. + dest_label must be at least 44 characters. */ +static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label, + char *dest_label, int nid) +{ + struct conexant_spec *spec = codec->spec; + int i; + + if (!spec->fixup_stereo_dmic) + return 0; + + for (i = 0; i < AUTO_CFG_MAX_INS; i++) { + int def_conf; + if (spec->autocfg.inputs[i].pin != nid) + continue; + + if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC) + return 0; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) + return 0; + + /* Finally found the inverted internal mic! */ + snprintf(dest_label, 44, "Inverted %s", label); + return 1; + } + return 0; +} + static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, const char *label, const char *pfx, int cidx) @@ -4216,14 +4247,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + char rightch_label[44]; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + /* Make two independent kcontrols for left and right */ + int err = cx_auto_add_volume_idx(codec, label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, 3); } return 0; } @@ -4236,9 +4278,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, int i, con; nid = spec->imux_info[idx].pin; - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { + char rightch_label[44]; + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + int err = cx_auto_add_volume_idx(codec, label, " Boost", + cidx, nid, HDA_INPUT, 0, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, " Boost", + cidx, nid, HDA_INPUT, 0, 2); + } return cx_auto_add_volume(codec, label, " Boost", cidx, nid, HDA_INPUT); + } con = __select_input_connection(codec, spec->imux_info[idx].adc, nid, &mux, false, 0); if (con < 0) @@ -4405,22 +4457,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4431,10 +4491,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4477,7 +4539,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds -- 1.7.9.1

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Re: [alsa-devel] [PATCH 5/6] sound/pci/hda: CX20549 doesn't need pin_amp_workaround.
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:35 +0200, Michael Karcher wrote: > > CX20549 (ctx5045) doesn't accept data on index 1 for output pins, > as shown in the following hda-var transaction: > > $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126 > nid = 0x10, verb = 0x300, param = 0xb126 > value = 0x0 > $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001 > nid = 0x10, verb = 0xb00, param = 0x8001 > value = 0x0 OK, it's good to know. Could you fix it in the corresponding part in patch_conexant_auto(), too? Takashi > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 8d7a977..3ab7b25 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -1087,7 +1087,6 @@ static int patch_cxt5045(struct hda_codec *codec) > if (!spec) > return -ENOMEM; > codec->spec = spec; > - codec->pin_amp_workaround = 1; > > spec->multiout.max_channels = 2; > spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); > -- > 1.7.9.1 > > >

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Re: [alsa-devel] [PATCH 4/6] sound/pci/hda: Remove CD control from model=benq for CX20549
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:25 +0200, Michael Karcher wrote: > > The ID used for detection of the BenQ R55E actually identifies the > Quana TW3 ODM design, which is also used for the Gigabyte W551 laptop > series. Schematics on the internet clearly indicate that the "Port C" > (analog input connected to record source #4 and mixer input #4) is > unconnected. Well, such an action needs to be done carefully since the documentation often lies. But I also doubt that there is any analog CD line. If this is confirmed that actually it's not present on the real machine, fine to remove it indeed. Takashi > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index bd7e7ce..8d7a977 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -695,7 +695,6 @@ static const struct hda_input_mux cxt5045_capture_source = { > static const struct hda_input_mux cxt5045_capture_source_benq = { > .num_items = 5, > .items = { > - { "CD", 0x4 }, > { "Internal Mic", 0x1 }, > { "Mic", 0x2 }, > { "Line In", 0x3 }, > @@ -820,9 +819,6 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { > - HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), > - HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), > - > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > -- > 1.7.9.1 > > >

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Re: [alsa-devel] [PATCH 3/6] sound/pci/hda: fix record volume controls of CX20459 ("Venice")
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:16 +0200, Michael Karcher wrote: > > The "input converter" widget of the CX20459 has only one input amplifier, > expose that one as "Capture Volume/Capture Switch". The actual record > source selection is already exposed through the separately installed > input mux. > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 7d92aac..bd7e7ce 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -798,10 +798,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, > } > > static const struct snd_kcontrol_new cxt5045_mixers[] = { > - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), > - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), > - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), > - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), > + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), > + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), > @@ -822,27 +820,18 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { > - HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT), > - HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT), > HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), > HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), > > - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), > - HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > - HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), > - > {} > }; > > static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = { > - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), > - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), > - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), > - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), > + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), > + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT), > @@ -978,16 +967,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { > .put = conexant_mux_enum_put, > }, > /* Audio input controls */ > - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), > - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), > - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), > - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), > - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), > - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), > + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), > { } /* end */ > }; Looks good. Takashi

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Re: [alsa-devel] [PATCH 2/6] sound/pci/hda: Rename capture sources of CX20549 to match common conventions
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:06 +0200, Michael Karcher wrote: > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 8c6523b..7d92aac 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -687,18 +687,18 @@ static const struct hda_channel_mode cxt5045_modes[1] = { > static const struct hda_input_mux cxt5045_capture_source = { > .num_items = 2, > .items = { > - { "IntMic", 0x1 }, > - { "ExtMic", 0x2 }, > + { "Internal Mic", 0x1 }, > + { "Mic", 0x2 }, > } > }; > > static const struct hda_input_mux cxt5045_capture_source_benq = { > .num_items = 5, > .items = { > - { "IntMic", 0x1 }, > - { "ExtMic", 0x2 }, > - { "LineIn", 0x3 }, > { "CD", 0x4 }, > + { "Internal Mic", 0x1 }, > + { "Mic", 0x2 }, > + { "Line In", 0x3 }, This could be simply "Line". Other than that, looks good. Takashi

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Re: [alsa-devel] [PATCH 1/6] sound/pci/hda: Fix Amp-In count for widgets with implied selector
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:29:56 +0200, Michael Karcher wrote: > > A selector widget with AC_WCAP_IN_AMP has multiple input amplifiers (if > the selector widget would have only one common amp, it would be labeled > as "output amplifier" instead). This is just like mixer widgets. > > If a non-mixer/selector widget has an input amplifier, it is only one > input amplifier, even if it has a connection list with more than one > entry. The implied selector is placed before the single input amplifier. > > At least that is how I read the HDA specs, and what Conexant implements in > their CX20549 codec. It's rather cx20549-specific behavior. If you take a look at the BIOS auto-parser, spec->single_adc_amp is set as a workaround. Yes, the proc file output should be fixed as well, but you can't change it globally for all codecs. Takashi > diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c > index 254ab52..9b942f0 100644 > --- a/sound/pci/hda/hda_proc.c > +++ b/sound/pci/hda/hda_proc.c > @@ -653,7 +653,7 @@ static void print_codec_info(struct snd_info_entry *entry, > snd_iprintf(buffer, " Amp-In vals: "); > print_amp_vals(buffer, codec, nid, HDA_INPUT, > wid_caps & AC_WCAP_STEREO, > - wid_type == AC_WID_PIN ? 1 : conn_len); > + (wid_type == AC_WID_AUD_SEL || wid_type == AC_WID_AUD_MIX) ? conn_len : 1); > } > if (wid_caps & AC_WCAP_OUT_AMP) { > snd_iprintf(buffer, " Amp-Out caps: "); > -- > 1.7.9.1 > > >

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Re: [alsa-devel] [PATCH 0/6] sound/pci/hda: CX20549 cleanup/fixes
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:29:37 +0200, Michael Karcher wrote: > > This series of patches fixes recording on a Gigabyte W551N laptop that > seems to be identical to the BenW R55E. The Audio Codec uses the same > subsystem ID, which is assigned to Quanta. The Gigabyte laptop is based > on the Quanta TW3 ODM. Thanks for the patch. I'll review and give comments back for each. > (for the whole series) > Signed-Off-By: Michael Karcher <kernel(a)mkarcher.dialup.fu-berlin.de> Please give your sign-off in each patch changelog, so that I can apply the patches directly via git-am. Also, the subject lines should be with ALSA: prefix. Take a look at other git commits to that file as an example. Takashi

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