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Re: [alsa-devel] HBR audio with intel HDA
by Anssi Hannula 02 Aug '13

02 Aug '13

02.08.2013 19:14, Ashecrow kirjoitti: > > Le 2 août 2013 à 15:42, Anssi Hannula a écrit : > >> 02.08.2013 16:11, Ashecrow kirjoitti: >>> Hi Anssi, >>> Thanks for your answer >>> >>> Test on kernel 3.8 + non-pcm bit in AES0: >>> aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif >>> => same result: no signal >> >> Hmh, OK. >> >>> With kernel upgrade to 3.9: same result >>> >>> >>> Note that I have tested with alsa* 1.025 and intel-hda driver included in kernel >>> >>> Do you have any other idea? >> >> You could try DEV=1 instead of DEV=0 (according to your log DEV=0 is >> correct, though), and truehd (thd.spdif from ffmpeg samples). >> >> If neither of those work, could you provide another alsa-info while >> playing back using AES0=0x06 (with alsa debug enabled again)? >> > I have already tried DEV=1 but it does the same. Moreover, let me note that I play DTS and dolby fine with DEV=0 in xbmc. So I guess it is the right device. OK. Note that in some cases you can hear stereo audio or AC-3/DTS even when using wrong DEV=x (but HBR or multichannel audio would not work). > see attached alsa-info.txt_dev0_kernel-3.8_aes0_dts-hd which is result of playing dts-hd spdif encapsulated with kernel 3.8 in debug mode and AES0=0x06 through command bellow > > aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif > > see attached alsa-info.txt_dev0_kernel-3.8_aes0_true-hd which is result of playing dts-hd spdif encapsulated with kernel 3.8 in debug mode and AES0=0x06 through command bellow > > aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 thd.spdif > Looks like your kernel is definitely too old to have working HBR on Intel (the "ASP channel x => slot y" messages should have x=y), so you'll need 3.9+. Unfortunately I don't see anything else wrong (and HBR+intel works for me), so I don't have anything else to suggest at the moment. Sorry. > Thanks for your help > >> >> >>> Thanks >>> >>> >>> Le 1 août 2013 à 18:37, Anssi Hannula a écrit : >>> >>>> 01.08.2013 19:29, ashecrow42(a)free.fr kirjoitti: >>>>> Hi there, >>>> >>>> Hi! >>>> >>>>> I have troubles using HBR audio (DTS-HD and Dolby TrueHD). With every >>>>> test, I get no signal in the AV receiver. I have tested using a dts-hd >>>>> encapsulated in spdif sample file (found there >>>>> http://samples.mplayerhq.hu/hdmi/) and commands bellow >>>>> >>>>> - aplay -D hdmi devices >>>>> - aplay -D hw devices >>>>> - aplay -D plughw devices >>>> >>>> You need to set the non-pcm bit in AES0. Use e.g. "-D hdmi:AES0=0x06". >>>> >>>>> Note that I use ubuntu raring with 3.8 kernel that includes Xingchao's >>>>> HBR kernel patches. >>>>> >>>>> >>>>> A. Configurations I have tested: >>>>> >>>>> >>>>> - hda driver upgrade (alsa devel ubuntu team daily builds) >>>>> >>>>> - upgrade alsa-lib & alsa-utils (alsa-testing launchpad) >>>>> >>>>> - kernel upgrade to 3.9 >>>> >>>> IIRC 3.9 (or newer) may indeed be needed since you have both hdmi and spdif. >>>> >>>>> - snd_hda_codec_realtek module backlist (to try to disable SPDIF) >>>>> /etc/modprobe.d/blacklist.conf >>>>> >>>>> - snd_hda_codec_realtek index = -2 in /etc/modprobe.d/alsa-base.conf >>>>> >>>>> - fresh new ubuntu raring install with no modifications at all: still >>>>> no signal >>>>> >>>>> B. Info: >>>>> >>>>> cat /proc/asound/card0/codec* | grep Codec >>>>> Codec: Realtek ALC892 >>>>> Codec: Intel CougarPoint HDMI >>>>> >>>>> I use zotac ZBOX ID82 (intel HM65 chipset with hdmi and spdif) >>>>> >>>>> with alsa debug and alsa debug verbose enabled in kernel I got the >>>>> alsa-info.txt attached file produced on raring with no modification >>>>> while playing dts-hd sample file with command: >>>>> >>>>> >>>>> aplay -D hdmi:CARD=PCH,DEV=0 -c8 -fs16_le -r192000 dts.spdif >>>> >>>> As per above you need to use >>>> aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif >>>> >>>>> >>>>> C. I add that: >>>>> >>>>> - In every test I have non HD audio protocols (DTS and dolby) working >>>>> well in the same configuration >>>>> >>>>> - I have tried with windows just to see if my hardware is broken and I >>>>> get DTS-HD audio with no problem >>>>> >>>>> - Takashi, I've read your HD-Audio.txt file and as far as I see, I have >>>>> no known error message in kernel log >>>>> >>>>> >>>>> => I and Wingchao Wang have conclued that it is driver or codec >>>>> related. Can you help me please? >>>>> >>>>> Thanks a lot >>>>> PS: sorry if you receive this email twice, I have sent it again as I >>>>> was not member of the list the first time. >>>> >>>> >>>> -- >>>> Anssi Hannula >>> >> >> >> -- >> Anssi Hannula > -- Anssi Hannula

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix empty list check in dapm_new_mux()
by Lars-Peter Clausen 02 Aug '13

02 Aug '13

list_first_entry() will always return a valid pointer, even if the list is empty. So the check whether path is NULL will always be false. So we end up calling dapm_create_or_share_mixmux_kcontrol() with a path struct that points right in the middle of the widget struct and by trying to modify the path the widgets memory will become corrupted. Fix this by using list_emtpy() to check if the widget doesn't have any paths. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This is the real fix for the rt5640 crash. --- sound/soc/soc-dapm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16..196cea7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -747,13 +747,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); if (ret < 0) return ret; -- 1.8.0

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[alsa-devel] HBR audio with intel HDA
by ashecrow42＠free.fr 02 Aug '13

02 Aug '13

Hi there, I have troubles using HBR audio (DTS-HD and Dolby TrueHD). With every test, I get no signal in the AV receiver. I have tested using a dts-hd encapsulated in spdif sample file (found there http://samples.mplayerhq.hu/hdmi/) and commands bellow - aplay -D hdmi devices - aplay -D hw devices - aplay -D plughw devices Note that I use ubuntu raring with 3.8 kernel that includes Xingchao's HBR kernel patches. A. Configurations I have tested: - hda driver upgrade (alsa devel ubuntu team daily builds) - upgrade alsa-lib & alsa-utils (alsa-testing launchpad) - kernel upgrade to 3.9 - snd_hda_codec_realtek module backlist (to try to disable SPDIF) /etc/modprobe.d/blacklist.conf - snd_hda_codec_realtek index = -2 in /etc/modprobe.d/alsa-base.conf - fresh new ubuntu raring install with no modifications at all: still no signal B. Info: cat /proc/asound/card0/codec* | grep Codec Codec: Realtek ALC892 Codec: Intel CougarPoint HDMI I use zotac ZBOX ID82 (intel HM65 chipset with hdmi and spdif) with alsa debug and alsa debug verbose enabled in kernel I got the alsa-info.txt attached file produced on raring with no modification while playing dts-hd sample file with command: aplay -D hdmi:CARD=PCH,DEV=0 -c8 -fs16_le -r192000 dts.spdif C. I add that: - In every test I have non HD audio protocols (DTS and dolby) working well in the same configuration - I have tried with windows just to see if my hardware is broken and I get DTS-HD audio with no problem - Takashi, I've read your HD-Audio.txt file and as far as I see, I have no known error message in kernel log => I and Wingchao Wang have conclued that it is driver or codec related. Can you help me please? Thanks a lot PS: sorry if you receive this email twice, I have sent it again as I was not member of the list the first time.

3 3

[alsa-devel] [PATCH 01/10] ASoC: dapm: Move snd_soc_dapm_update from dapm context to card
by Lars-Peter Clausen 02 Aug '13

02 Aug '13

The update field of a DAPM context is only assigned while the card's dapm_mutex is locked, the field is also cleared again while the mutex is stil locked. So there will only ever be one DAPM context at a time with a non-NULL update field. So it is safe to move the update field from the DAPM context struct to the card struct. Doing so will allow further cleanups in this area. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 2 -- include/sound/soc.h | 1 + sound/soc/soc-dapm.c | 22 +++++++++++----------- 3 files changed, 12 insertions(+), 13 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e77c6f5..3397292 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -575,8 +575,6 @@ struct snd_soc_dapm_context { struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - struct snd_soc_dapm_update *update; - void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7..b1e1f96 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1042,6 +1042,7 @@ struct snd_soc_card { /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; struct snd_soc_dapm_stats dapm_stats; + struct snd_soc_dapm_update *update; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 366daef..7449e27 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1425,7 +1425,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, static void dapm_widget_update(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_update *update = dapm->update; + struct snd_soc_dapm_update *update = dapm->card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; unsigned int wi; @@ -1959,9 +1959,9 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2002,9 +2002,9 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2693,11 +2693,11 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2775,11 +2775,11 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2928,11 +2928,11 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); -- 1.8.0

2 15

[alsa-devel] [PATCH] ALSA: hda - Fix missing fixup for Mac Mini with STAC9221
by Takashi Iwai 02 Aug '13

02 Aug '13

A fixup for Apple Mac Mini was lost during the adaption to the generic parser because the fallback for the generic ID 8384:7680 was dropped, and it resulted in the silence output (and maybe other problems). Unfortunately, just adding the missing subsystem ID wasn't enough, in this case. The subsystem ID of this machine is 0000:0100 (what Apple thought...?), and since snd_hda_pick_fixup() doesn't take the vendor id zero into account, the driver ignored this entry. Now it's fixed to regard the vendor id zero as a valid value. Reported-and-tested-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46..48a9d00 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 92b9b43..6d1924c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2819,6 +2819,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm: Check return value of snd_soc_cnew()
by Lars-Peter Clausen 01 Aug '13

01 Aug '13

snd_soc_cnew() can return NULL, so we should check the result before trying to use it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d74c356..b4fae87 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -671,8 +671,10 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); - kcontrol->private_free = dapm_kcontrol_free; kfree(long_name); + if (!kcontrol) + return -ENOMEM; + kcontrol->private_free = dapm_kcontrol_free; ret = dapm_kcontrol_data_alloc(w, kcontrol); if (ret) { -- 1.8.0

2 4

[alsa-devel] [PATCH] ASoC: rt5640: remove unused mux
by bardliao＠realtek.com 01 Aug '13

01 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Remove unused "INL Mux" and "INR Mux". Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- I meet a kernel panic issue if I don't remove the unused widget and modify something in rt5640_dapm_widgets[] and/or rt5640_dapm_routes[]. The message is "Unable to handle kernel paging request at virtual address 6b6b6bd3". PC is at dapm_clear_walk_output+0x18/0x7c LR is at dapm_clear_walk_output+0x18/0x7c If I remove those unused widget, ie. not appear in rt5640_dapm_routes[], everything is fine. So I think they should be removed. --- sound/soc/codecs/rt5640.c | 26 -------------------------- 1 file changed, 26 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..4db7314 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -737,29 +737,6 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; -/* INL/R source */ -static const char * const rt5640_inl_src[] = { - "IN2P", "MONOP" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inl_enum, RT5640_INL_INR_VOL, - RT5640_INL_SEL_SFT, rt5640_inl_src); - -static const struct snd_kcontrol_new rt5640_inl_mux = - SOC_DAPM_ENUM("INL source", rt5640_inl_enum); - -static const char * const rt5640_inr_src[] = { - "IN2N", "MONON" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inr_enum, RT5640_INL_INR_VOL, - RT5640_INR_SEL_SFT, rt5640_inr_src); - -static const struct snd_kcontrol_new rt5640_inr_mux = - SOC_DAPM_ENUM("INR source", rt5640_inr_enum); - /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -1005,9 +982,6 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_IN_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL, RT5640_PWR_IN_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux), - SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0, rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)), -- 1.8.1.1.439.g50a6b54

4 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Sanitize parameter passing
by Dimitris Papastamos 01 Aug '13

01 Aug '13

No need to hold on to the `codec' pointer. We can use the `dsp' pointer and grab all the information we need from there. This makes the parameters for the functions a bit more sane and idiomatic. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 31 ++++++++++++++++--------------- sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 17 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3168224..b38f350 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,7 +227,6 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; - struct snd_soc_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -484,7 +483,7 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) struct snd_kcontrol_new *kcontrol; int ret; - if (!ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -498,14 +497,14 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = snd_soc_add_card_controls(ctl->card, + ret = snd_soc_add_card_controls(adsp->card, kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - ctl->kcontrol = snd_soc_card_get_kcontrol(ctl->card, + ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card, ctl->name); list_add(&ctl->list, &adsp->ctl_list); @@ -777,11 +776,10 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } -static int wm_adsp_create_control(struct snd_soc_codec *codec, +static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *region) { - struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); struct wm_coeff_ctl *ctl; struct wmfw_ctl_work *ctl_work; char *name; @@ -840,7 +838,6 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card; ctl->adsp = dsp; ctl->len = region->len; @@ -877,7 +874,7 @@ err_name: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +static int wm_adsp_setup_algs(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -1065,7 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1082,7 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1111,7 +1108,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1128,7 +1125,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1145,7 +1142,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1365,6 +1362,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1399,7 +1398,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; @@ -1492,6 +1491,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: /* @@ -1554,7 +1555,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 64087fb..d018dea 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -39,6 +39,7 @@ struct wm_adsp { int type; struct device *dev; struct regmap *regmap; + struct snd_soc_card *card; int base; int sysclk_reg; -- 1.8.3.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: Make secondary I2S DAI device a child of primary
by Mark Brown 01 Aug '13

01 Aug '13

From: Mark Brown <broonie(a)linaro.org> More for neatness than for any great utility. Really we shouldn't be creating the child device at all, refactoring will follow. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a3e4b49..47e08dd 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1019,6 +1019,8 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) if (IS_ERR(i2s->pdev)) return NULL; + i2s->pdev->dev.parent = &pdev->dev; + platform_set_drvdata(i2s->pdev, i2s); ret = platform_device_add(i2s->pdev); if (ret < 0) -- 1.8.3.2

1 0

[alsa-devel] How do I get this patch?
by Christian Winger Nyman 01 Aug '13

01 Aug '13

I have the VB-99, so this patch would be fantastic. But I don't understand, will it be implemented in the Linux 3.10 kernel or what would be the easiest way for me to get it? I run Ubuntu Studio 13.04. Thanks for any guidance. http://comments.gmane.org/gmane.linux.alsa.devel/109660

2 1

[alsa-devel] [PATCH v2] ASoC: Add PCM1681 codec driver.
by Marek Belisko 01 Aug '13

01 Aug '13

PCM1681 can be controlled via I2C, SPI or in bootstrap mode (no control mode). This code add support only for I2C mode. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- Changes from v1: - fix de-emphasis handling (disable when invalid freq is used) - constify deemph freq - digital_mute refactoring - remove empty functions (probe and remove) - call set_deemph in hw_params - remove CONFIG_I2C - small refactoring - remove PCM1681 from controls .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 313 ++++++++++++++++++++ 4 files changed, 334 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 0000000..4df1718 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d038a58..8ace02a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -297,6 +298,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fab4086..d8fa18e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -43,6 +43,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -173,6 +174,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..27da41b --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,313 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko(a)streamunlimited.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static const int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, enable = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + if (val != -1) { + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + enable = 1; + } else + enable = 0; + + /* enable/disable deemphasis functionality */ + return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, enable); +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + else + val = 0; + + return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + if (ret < 0) + return ret; + + return pcm1681_set_deemph(codec); +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko(a)streamunlimited.com>"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 1

[alsa-devel] ASoC fixes for v3.11
by Mark Brown 01 Aug '13

01 Aug '13

The following changes since commit 5ae90d8e467e625e447000cb4335c4db973b1095: Linux 3.11-rc3 (2013-07-28 20:53:33 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc3 for you to fetch changes up to 3fef7f795fff7ccc58d55a28315ca73305515884: Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus (2013-07-31 21:07:23 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.11 A fix to make sure userspace knows when control writes have caused a change in value, fixing some UIs, plus a few few driver fixes mainly cleaning up issues from recent refactorings on less mainstream platforms. ---------------------------------------------------------------- Dimitris Papastamos (1): ASoC: wm0010: Fix resource leak Lars-Peter Clausen (2): ASoC: dapm: Fix return value of snd_soc_dapm_put_{volsw,enum_virt}() ASoC: bf5xx-ac97: Fix compile error with SND_BF5XX_HAVE_COLD_RESET Mark Brown (4): ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring Merge remote-tracking branch 'asoc/fix/blackfin' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus Ralf Baechle (1): ASoC: au1x: Fix build sound/soc/au1x/ac97c.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- sound/soc/blackfin/bf5xx-ac97.h | 1 - sound/soc/codecs/wm0010.c | 24 ++++++++++++------------ sound/soc/soc-dapm.c | 5 ++--- 5 files changed, 18 insertions(+), 19 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: wm0010: Use DMA-safe memory for SPI transfers
by Dimitris Papastamos 01 Aug '13

01 Aug '13

We should be allocating our buffers for the SPI transfers from the DMA zone. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 10adc41..d5ebcb0 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -420,7 +420,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) xfer->codec = codec; list_add_tail(&xfer->list, &xfer_list); - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); @@ -429,7 +429,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) } xfer->t.rx_buf = out; - img = kzalloc(len, GFP_KERNEL); + img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); @@ -523,14 +523,14 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL); + img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } - out = kzalloc(fw->size, GFP_KERNEL); + out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; @@ -670,14 +670,14 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = -ENOMEM; len = pll_rec.length + 8; - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); goto abort; } - img_swap = kzalloc(len, GFP_KERNEL); + img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); -- 1.8.3.4

2 1

[alsa-devel] [PATCH 0/4] ALSA: Implement core jack support for kcontrol
by Felipe F. Tonello 01 Aug '13

01 Aug '13

From: "Felipe F. Tonello" <eu(a)felipetonello.com> This patch series enable KControl support for core Jack and update current codecs. This will will enable moderns user-space daemons, such as PulseAudio(>=2.0), to have jack detection support via newer ALSA KControl interface. Patch 1: Actually implementation using KControl. Patch 2: It also ensures HDA drivers to use properly this new ALSA Jack implementation. Patches 3-4: This patch makes sure that all users of ALSA Jack uses this new standard way to report detections to user-space. Felipe F. Tonello (4): ALSA: Added jack detection kcontrol support ALSA: HDA: Updating jack implementation according new ALSA Jacks ALSA: SoC: Updating jack implementation according new ALSA Jacks ALSA: oxygen: Updating jack implementation according new ALSA Jacks include/sound/jack.h | 6 ++++-- include/sound/soc.h | 2 +- sound/core/Kconfig | 1 + sound/core/ctljack.c | 3 ++- sound/core/jack.c | 29 +++++++++++++++++++++++++++-- sound/pci/hda/Kconfig | 8 -------- sound/pci/hda/hda_codec.h | 2 -- sound/pci/hda/hda_jack.c | 38 +++++++++++++++++--------------------- sound/pci/hda/hda_jack.h | 4 +--- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 4 ++-- sound/soc/mid-x86/mfld_machine.c | 6 +++--- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 4 ++-- sound/soc/omap/omap-twl4030.c | 4 ++-- sound/soc/omap/rx51.c | 6 +++--- sound/soc/pxa/hx4700.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/ttc-dkb.c | 8 ++++---- sound/soc/pxa/z2.c | 4 ++-- sound/soc/samsung/goni_wm8994.c | 4 ++-- sound/soc/samsung/h1940_uda1380.c | 4 ++-- sound/soc/samsung/littlemill.c | 10 +++++----- sound/soc/samsung/lowland.c | 6 +++--- sound/soc/samsung/rx1950_uda1380.c | 4 ++-- sound/soc/samsung/smartq_wm8987.c | 4 ++-- sound/soc/samsung/speyside.c | 6 +++--- sound/soc/samsung/tobermory.c | 4 ++-- sound/soc/soc-jack.c | 4 ++-- sound/soc/tegra/tegra_alc5632.c | 4 ++-- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 8 ++++---- 32 files changed, 107 insertions(+), 94 deletions(-) -- 1.8.1.4

6 30

[alsa-devel] [patch] ASoC: dapm: using freed pointer in dapm_kcontrol_add_widget()
by Dan Carpenter 01 Aug '13

01 Aug '13

There is a typo here so we end up using the old freed pointer instead of the newly allocated one. (If the "n" is zero then the code works, obviously). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Only needed in linux-next diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9abb3b2..d74c356 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -225,13 +225,13 @@ static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, new_data = krealloc(data, sizeof(*data) + sizeof(widget) * n, GFP_KERNEL); - if (!data) + if (!new_data) return -ENOMEM; - data->wlist.widgets[n - 1] = widget; - data->wlist.num_widgets = n; + new_data->wlist.widgets[n - 1] = widget; + new_data->wlist.num_widgets = n; - kcontrol->private_data = data; + kcontrol->private_data = new_data; return 0; }

4 7

[alsa-devel] New dapm panic on beaver with next-20130730
by Olof Johansson 31 Jul '13

31 Jul '13

Hi Mark, Lars-Peter, I noticed the below panic on beaver (Tegra30). Seems like seaboard (Tegra20) is fine. This is with next-20130730, tegra_defconfig. I noticed a bunch of recent dapm changes in the asoc tree on the 29th, and I was able to bisect it down to: commit 5106b92f80a2cd37c52cffed80b4f5acfb77ccfd Author: Lars-Peter Clausen <lars(a)metafoo.de> Date: Mon Jul 29 17:14:00 2013 +0200 ASoC: dapm: Keep a list of paths per kcontrol Currently we store for each path which control (if any at all) is associated with that control. But we are only ever interested in the reverse relationship, i.e. we want to know all the paths a certain control is associated with. This is currently implemented by always iterating over all paths. This patch updates the code to keep a list for each control which contains all the paths that are associated with that control. This improves the run time of e.g. soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n being the number of paths for the card) to O(1). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)linaro.org> Panic output is below. I need to figure out why I'm not actually getting a backtrace in panics any more as well so this is of somewhat limited use, unfortuantely. Running a brute force function lookup on all words on the stack gives a rough idea of call path though: c03b1bb4 dapm_clear_walk_output c03b3450 dapm_generic_check_power c03b56b0 dapm_power_widgets c03b608c snd_soc_dapm_new_widgets [c069f8f8 kallsyms_token_index] c03ae9fc soc_post_component_init c03b0dd0 soc_probe_link_dais c069abb0 kallsyms_token_index [c029ac24 devm_kzalloc_release] c03b1870 snd_soc_register_card c03c3964 tegra_rt5640_probe c0707b74 tegra_rt5640_driver_init c02999e0 platform_drv_probe c02999c8 platform_drv_probe c029878c driver_probe_device [ 1.845620] Unable to handle kernel paging request at virtual address 6b6b6bd3 [ 1.852855] pgd = c0004000 [ 1.855552] [6b6b6bd3] *pgd=00000000 [ 1.859125] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.864422] Modules linked in: [ 1.867473] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-rc3-next-20130730 #42 [ 1.875023] task: ef064a40 ti: ef074000 task.ti: ef074000 [ 1.880421] PC is at dapm_clear_walk_output+0x8/0x68 [ 1.885371] LR is at dapm_clear_walk_output+0x54/0x68 [ 1.890408] pc : [<c03bb77c>] lr : [<c03bb7c8>] psr: 20000113 [ 1.890408] sp : ef075ce8 ip : 00000000 fp : 00000010 [ 1.901858] r10: ef075d2c r9 : c076dd00 r8 : c076dbf8 [ 1.907066] r7 : ef075d34 r6 : eeb490b8 r5 : 6b6b6bd3 r4 : ef39c4b0 [ 1.913574] r3 : 00000069 r2 : 00000192 r1 : 6b6b6bd3 r0 : eeb490b8 [ 1.920084] Flags: nzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 1.927372] Control: 10c5387d Table: 8000404a DAC: 00000015 [ 1.933101] Process swapper/0 (pid: 1, stack limit = 0xef074238) [ 1.939090] Stack: (0xef075ce8 to 0xef076000) [ 1.943434] 5ce0: ef39c4b0 ef3ad728 eeb490b8 c03bb7c8 ef3ad6c0 00000000 [ 1.951591] 5d00: 00000000 c03bd074 c076dcf8 ef3ad6c0 ef075d3c c03bf310 00000000 c076dcf8 [ 1.959747] 5d20: 00000000 eec537f0 c077ca0c ef075d2c ef075d2c ef075d34 ef075d34 ef075d3c [ 1.967904] 5d40: ef075d3c c01c298c eec66734 ef3ad584 c076dcd4 000000f0 ef3ad5a0 c076dce8 [ 1.976060] 5d60: c0587888 00000005 00000010 c03bfcd8 c0587888 00000000 c076dbf8 c076dc34 [ 1.984216] 5d80: 00000000 eeb56010 eeb49000 c076dbf8 00000000 c06ab0f8 c076dda8 00000000 [ 1.992374] 5da0: 00000002 c03b8618 00000000 00000000 00000000 eeb56010 00000000 ef21e3c0 [ 2.000531] 5dc0: c076dbf8 c076dda8 ef21e0c0 00000000 00000002 c03ba9e4 c076dc08 000000d0 [ 2.008688] 5de0: ef000500 c076a5a8 c076dd08 00000000 ef21e184 c076dcf8 c076a5ec eeb49000 [ 2.016844] 5e00: c076dce0 c076a5a8 ef3b1e8c eeb49034 c06a63ac c076dc18 c02a2a24 c076dbf8 [ 2.025001] 5e20: 000005d4 ef369a90 c17dd4f4 00000000 000000b3 ef074028 00000000 c03bb484 [ 2.033157] 5e40: c17dba28 c076db24 ef18ae10 c03cd59c c07bffd8 ef18ae10 00000000 c076db38 [ 2.041315] 5e60: c0712a54 c02a17e0 c02a17c8 c02a058c 00000000 ef18ae10 c076db38 ef18ae44 [ 2.049471] 5e80: 00000000 c02a0778 00000000 c076db38 c02a06ec c029eac8 ef06d0e0 ef181d74 [ 2.057627] 5ea0: c076db38 ef3af840 c07526d8 c029fd44 c06ab0d4 c076db38 00000006 c076db38 [ 2.065783] 5ec0: 00000006 c0777a40 c0777a40 c02a0d54 c07277c0 00000006 c0777a40 c0777a40 [ 2.073941] 5ee0: c0712a54 c00089f8 ef025900 c0659cf4 ef1160c0 c053b498 00000000 00000001 [ 2.082098] 5f00: 0000027c c073bd78 c17dd77b c0550080 000000b3 c003d884 00000000 c071d9dc [ 2.090254] 5f20: c06937c8 c06ed458 00000006 00000006 c003d198 c003d1f0 00000000 c07277c0 [ 2.098411] 5f40: 00000006 c0777a40 c0777a40 c06f027c 000000b3 c071d9dc c071d9d0 c06f0970 [ 2.106569] 5f60: 00000006 00000006 c06f027c ffffbfc7 ff7ffffd fffff7ff 9fffffff 00000000 [ 2.114726] 5f80: c0526f58 00000000 00000000 00000000 00000000 00000000 00000000 c0526f60 [ 2.122884] 5fa0: 00000000 00000000 c0526f58 c000e578 00000000 00000000 00000000 00000000 [ 2.131040] 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 2.139198] 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 bf7fcdff eefffeff [ 2.147359] Code: 1affffe1 eafffff5 e92d4070 e1a05001 (e5914000) [ 2.153588] ---[ end trace a5214c5b7e3cefc9 ]--- [ 2.158220] Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b

4 5

[alsa-devel] [PATCH v2] ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot)
by Peter Ujfalusi 31 Jul '13

31 Jul '13

Just recently OMAP4 legacy boot support has been removed. No reason to keep the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5 can boot in this mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Changes since v1: - Remove the header file which is no longer in use. Regards, Peter include/linux/platform_data/omap-abe-twl6040.h | 49 --------- sound/soc/omap/omap-abe-twl6040.c | 133 ++++++++----------------- 2 files changed, 41 insertions(+), 141 deletions(-) delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h deleted file mode 100644 index 5d298ac..0000000 --- a/include/linux/platform_data/omap-abe-twl6040.h +++ /dev/null @@ -1,49 +0,0 @@ -/** - * omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header. - * - * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com - * All rights reserved. - * - * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef _OMAP_ABE_TWL6040_H_ -#define _OMAP_ABE_TWL6040_H_ - -/* To select if only one channel is connected in a stereo port */ -#define ABE_TWL6040_LEFT (1 << 0) -#define ABE_TWL6040_RIGHT (1 << 1) - -struct omap_abe_twl6040_data { - char *card_name; - /* Feature flags for connected audio pins */ - u8 has_hs; - u8 has_hf; - bool has_ep; - u8 has_aux; - u8 has_vibra; - bool has_dmic; - bool has_hsmic; - bool has_mainmic; - bool has_submic; - u8 has_afm; - /* Other features */ - bool jack_detection; /* board can detect jack events */ - int mclk_freq; /* MCLK frequency speed for twl6040 */ -}; - -#endif /* _OMAP_ABE_TWL6040_H_ */ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7..ebb1390 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/mfd/twl6040.h> -#include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> #include <linux/of.h> @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/3] ASoC: Add Freescale i.MX S/PDIF controller driver
by Nicolin Chen 31 Jul '13

31 Jul '13

* This series of patches add i.MX S/PDIF drivers, including CPU DAI, machine. * It also add two missing sample rates support for spdif dummy codec drivers Nicolin Chen (3): ASoC: codec: spdif: Add S20_3LE and S24_LE support for dummy codec drivers ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 63 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/codecs/spdif_receiver.c | 2 + sound/soc/codecs/spdif_transmitter.c | 5 +- sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1276 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 233 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 9 files changed, 1758 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 7

Re: [alsa-devel] [PATCH -next] ASoC: rcar: fix return value check in rsnd_gen1_probe()
by Mark Brown 31 Jul '13

31 Jul '13

On Tue, Jul 30, 2013 at 07:51:37AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > In case of error, the function devm_ioremap_resource() returns ERR_PTR() > and never returns NULL. The NULL test in the return value check should be > replaced with IS_ERR(), and also remove the dev_err call to avoid redundant > error message. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: fsl: Fix module build
by Fabio Estevam 31 Jul '13

31 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Building imx_v6_v7_defconfig with all audio drivers as modules results in the folowing build error: ERROR: "imx_pcm_fiq_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_fiq_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined! ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined! Fix this by allowing SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to be also built as modules and by using 'IS_ENABLED' to cover the module case. Reported-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 4 ++-- sound/soc/fsl/imx-pcm.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3a79d01..c26449b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -109,11 +109,11 @@ config SND_SOC_IMX_SSI tristate config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_IMX_AUDMUX diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 9136625..5d5b733 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -38,7 +38,7 @@ struct imx_pcm_fiq_params { struct snd_dmaengine_dai_dma_data *dma_params_tx; }; -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); #else @@ -52,7 +52,7 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) } #endif -#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev, struct imx_pcm_fiq_params *params); void imx_pcm_fiq_exit(struct platform_device *pdev); -- 1.8.1.2

3 4

[alsa-devel] [PATCH v2] ASoC: wm0010: Fix resource leak
by Dimitris Papastamos 31 Jul '13

31 Jul '13

If kzalloc() fails for `img' then we are going to leak the memory for `out'. We are freeing the memory of all the tx/rx transfers but the tx/rx buf pointers will be NULL if we drop out earlier. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f5e8356..10adc41 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,6 +410,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + out = kzalloc(len, GFP_KERNEL); if (!out) { dev_err(codec->dev, @@ -417,6 +427,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; img = kzalloc(len, GFP_KERNEL); if (!img) { @@ -425,24 +436,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - - xfer->codec = codec; - list_add_tail(&xfer->list, &xfer_list); - spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; -- 1.8.3.4

2 1

[alsa-devel] [PATCH] ASoC: core: No need to use strncmp()
by Dimitris Papastamos 31 Jul '13

31 Jul '13

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d68a486..9f60894 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2309,7 +2309,7 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, return NULL; list_for_each_entry(kctl, &card->controls, list) - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + if (!strcmp(kctl->id.name, name)) return kctl; return NULL; } @@ -3056,7 +3056,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, return -EINVAL; list_for_each_entry(kctl, &card->controls, list) { - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + if (!strcmp(kctl->id.name, name)) { found = 1; break; } -- 1.8.3.4

2 5

[alsa-devel] [PATCH] ASoC: wm0010: Fix resource leak
by Dimitris Papastamos 31 Jul '13

31 Jul '13

This fixes "https://bugzilla.kernel.org/show_bug.cgi?id=60650". Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f5e8356..285026c 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,6 +410,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL | GFP_DMA); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort1; + } + out = kzalloc(len, GFP_KERNEL); if (!out) { dev_err(codec->dev, @@ -417,6 +424,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; img = kzalloc(len, GFP_KERNEL); if (!img) { @@ -425,24 +433,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - xfer->codec = codec; list_add_tail(&xfer->list, &xfer_list); spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; -- 1.8.3.4

2 3

[alsa-devel] [PATCH] SOUND: au1x: Fix build
by Ralf Baechle 31 Jul '13

31 Jul '13

d8b51c11ff5a70244753ba60abfd47088cf4dcd4 [ASoC: ac97c: Use module_platform_driver()] broke the build: CC sound/soc/au1x/ac97c.o /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__initcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__exitcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:334:31: warning: ‘au1xac97c_driver’ defined but not used [-Wunused-variable] make[5]: *** [sound/soc/au1x/ac97c.o] Error 1 make[4]: *** [sound/soc/au1x] Error 2 make[3]: *** [sound/soc] Error 2 Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");

2 1

[alsa-devel] [PATCH 1/1] conf/cards: add Audio Advantage Micro II
by Przemek Rudy 31 Jul '13

31 Jul '13

With related kernel chnages add extended suport for: - AES bits controls (via device suffix AESn= or tools like iecset) - SPDIF switch control, turning on/off the optical transmitter (via alsamixer tool) Signed-off-by: Przemek Rudy <prudy1(a)o2.pl> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 0d93175..8a6d9ca 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -101,6 +101,41 @@ USB-Audio."Audiophile USB (tm)".pcm.iec958 { } } +# For this card we can (and must to get IEC61937) set AES bits +USB-Audio."MicroII".pcm.iec958 "cards.USB-Audio.Audio Advantage MicroII.pcm.iec958" +USB-Audio."Audio Advantage MicroII".pcm.iec958 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + + type hooks + slave.pcm { + type hw + card $CARD + } + + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + preserve true + optional true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + { + name "IEC958 Playback Switch" + lock true + preserve true + value 1 + } + ] + } +} ################################################################################ -- 1.8.3.1

2 1

Re: [alsa-devel] [PATCH - Preliminary support for Xonar Xense (resubmitAdded support for Xonar Xense. Only speaker out and mic in working.
by Clemens Ladisch 30 Jul '13

30 Jul '13

Ian Dawes wrote: > Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Ian Dawes wrote: >>> Clemens Ladisch <clemens(a)ladisch.de> wrote: >>>> Ian Dawes wrote: >>>>> >>>>> Signed-off-by: Ian Dawes <madeallup.gen(a)gmail.com> >>>> >>>> The commit message was squashed into the subject line. There's >>>> probably an empty line missing between them. >>> >>> I'm not sure what this means. >> >> Did you intend for the subject to be this long? > > Probably not. I added "resubmitted" to the subject And the "Added support ..." was supposed to be in the subject, too? > How should I rectify this? Ensure there is an empty line after the subject. Regards, Clemens

2 2

Re: [alsa-devel] [PATCH 7/9] ASoC: tegra: move to generic DMA DT binding
by Stephen Warren 30 Jul '13

30 Jul '13

On 07/30/2013 04:12 AM, Richard Zhao wrote: > On Sat, Jul 27, 2013 at 03:58:21AM +0800, Stephen Warren wrote: >> On 07/23/2013 10:10 PM, Richard Zhao wrote: >>> - add tegra_dma_filter_data to specify dma info >>> DMA DT binding needs the device that raise dma request and dma name >>> to request a dma channel. tegra30_i2s is a special case. It should be ahub >>> device and it also has dma name that cannot handled by ASoC dmaengine code. >>> So we pass the info using filter data in snd_dmaengine_dai_dma_data. >>> - change i2s/ac97 drivers to use generic DT binding >>> - tegra30_i2s: alloc ahub tx/rx fifo at driver probe time >> >>> diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c >> >> I don't understand why the FIFO alloc/free had to move from >> startup()/shutdown() to probe()/remove(). > > The call path is: > [<c0520e38>] (dump_stack+0x78/0xbc) from [<c03bb854>] (tegra_pcm_request_chan+0x10/0x34) > [<c03bb854>] (tegra_pcm_request_chan+0x10/0x34) from [<c03b6c2c>] (dmaengine_pcm_new+0xe0/0x134) > [<c03b6c2c>] (dmaengine_pcm_new+0xe0/0x134) from [<c03b5824>] (soc_new_pcm+0x2f0/0x3cc) > [<c03b5824>] (soc_new_pcm+0x2f0/0x3cc) from [<c03aa1a0>] (soc_probe_link_dais+0x358/0x39c) > [<c03aa1a0>] (soc_probe_link_dais+0x358/0x39c) from [<c03ab334>] (snd_soc_instantiate_card+0x35c/0x7a0) > [<c03ab334>] (snd_soc_instantiate_card+0x35c/0x7a0) from [<c03ab9dc>] (snd_soc_register_card+0x264/0x330) > [<c03ab9dc>] (snd_soc_register_card+0x264/0x330) from [<c03be350>] (tegra_rt5640_probe+0xf4/0x17c) > [<c03be350>] (tegra_rt5640_probe+0xf4/0x17c) from [<c0294290>] (platform_drv_probe+0x18/0x1c) > > So I need to request dma channel at probe time. The AHUB driver should look up and cache the DMA channel information during its probe() call, and defer probe() there if necessary. The FIFO alloc/free calls should simply use the cached information for the allocated channel. That way, the changes are isolated in the AHUB driver and shouldn't require any changes in the I2S driver. The DT parsing and DMA request lookup will then happen well before the snd_soc_instantiate_card() backtrace you mention above, since it will prevent the card from being instantiated until all the ASoC components are fully probed. >> This will prevent later >> changes from making the connectivity between the AHUB FIFOs and I2S >> modules more dynamic, which is kind of the whole point of the AHUB HW >> module. > > That's because it's i2s driver to register pcm device rather not ahub. > Ideally dma channel is only bound to ahub. Long-term, this should change. However, I believe if you make the changes I mention above, this won't be an issue. >>> diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h >> >>> +#define TEGRA_DMA_NAME_MAX_LEN 10 >>> + >>> +/* >>> + * Generic DMA DT binding needs the device that raise dma request and dma name >>> + * to request a dma channel. tegra30_i2s is a special case. It should be ahub >>> + * device and it also has dma name that cannot handled by ASoC dmaengine code. >>> + * So we pass the info using filter data in snd_dmaengine_dai_dma_data. >>> + */ >>> +struct tegra_dma_filter_data { >>> + struct device *dma_dev; >>> + char dma_name[TEGRA_DMA_NAME_MAX_LEN]; >>> +}; >> >> I agree with Lars-Peter's comment that it'd be better to enhance the >> ASoC dmaengine support to allow the channel name to be specified so we >> don't need to much custom code. > > Not only dma-name but also dma client device for this case. > >> And that should use "char *" not a >> fixed-length array. > > Ok, then there' two solutions. > > struct tegra30_i2s { > ... > struct snd_dmaengine_dai_dma_data capture_dma_data; > char dma_rx_name[10]; > struct tegra_dma_filter_data filter_data_rx; > ... > struct snd_dmaengine_dai_dma_data playback_dma_data; > char dma_tx_name[10]; > struct tegra_dma_filter_data filter_data_tx; > }; > > probe: i2s->filter_data_rx.dma_name = i2s.dma_rx_name > > or > > probe: i2s->filter_data_rx.dma_name = kasprintf(GFP_KERNEL, "channel%d" ...); > remove: kfree(i2s->filter_data_rx.dma.name) > > Which one do you like? Either is fine, although the first option has less dynamic allocation, so is probably better. The AHUB driver should be constructing the channel names though.

1 0

[alsa-devel] [PATCH 1/2] ASoC: bf5xx-ac97: Fix compile error
by Lars-Peter Clausen 30 Jul '13

30 Jul '13

If CONFIG_SND_BF5XX_HAVE_COLD_RESET is enabled building the blackfin ac97 driver fails with the following compile error: sound/soc/blackfin/bf5xx-ac97.c: In function ‘asoc_bfin_ac97_probe’: sound/soc/blackfin/bf5xx-ac97.c:297: error: expected ‘;’ before ‘{’ token sound/soc/blackfin/bf5xx-ac97.c:302: error: label ‘gpio_err’ used but not defined The issue was introduced in commit 6dab2fd7 ("ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index efb1dae..e82eb37 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /* Request PB3 as reset pin */ ret = devm_gpio_request_one(&pdev->dev, CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); + if (ret) { dev_err(&pdev->dev, "Failed to request GPIO_%d for reset: %d\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - goto gpio_err; + return ret; } #endif -- 1.8.0

2 5

[alsa-devel] [PATCH v6 0/8] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 30 Jul '13

30 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) As the at91 chips are on their way to the common clock framework, I thought it may be better to retrieve the codec MCLK via this framework. So it needs Boris Brezillon patch set "ARM: at91: move to common clk framework" I've separated this patch because I don't know what is better: * applying the patchset with a hard coded clock, and once at91 has moved to common clock framework, update the code and change the DT binding. * wait for the common clock framework. Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. Maybe atmel_wm8731.c ? Since it could be merged with sam9g20_wm8731.c when the g20 is DT only (There will be some more work though). It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130729 + "ARM: at91: move to common clk framework" Changes from v5: * Update the wm8731 DT binding documentation with available audio endpoints. * Remove Uwe's signed-off since it seems to be "an integration signed-off" * Remove rate constrains if freq==0 in wm8731. * Allocate dynamically card, dai and private structures. * Add a separate patch to set the oscillator value via DT. NB: this last patch needs Boris Brezillon patch set "ARM: at91: move to common clk framework" Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Richard Genoud (7): ASoC: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration ASoC: sam9x5: get codec MCLK via device tree .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 39 ++++ Documentation/devicetree/bindings/sound/wm8731.txt | 9 + arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 32 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 218 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 60 +++++- 9 files changed, 392 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: ASoC: uda134x: fix codec driver
by Russell King 30 Jul '13

30 Jul '13

For some reason, the DAC/ADCs are not being powered up when I try and use the UDA1341 driver; this used to work. Looking back in the git history, I don't see anything obvious which would cause this regression. However, from dumping the register writes, it seems that the codec is powered down, and nothing calls set_bias_level to wake the codec up. Moreover, this driver hasn't had DAPM support added to it, which prevents platform drivers from taking advantage of DAPMs facilities. So, let's add DAPM support to the driver. As we move the power control for the DAC/ADC into DAPM, we no longer need it in set_bias_level() - this function just becomes a way to manipulate the power control and sync the register cache with the hardware at the appropriate point. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- I noticed this while checking the Assabet audio, which uses the UDA1341. I've also added the corresponding support to my Assabet audio driver. If this looks like the correct thing to do to fix this, please apply. sound/soc/codecs/uda134x.c | 88 ++++++++++++++++++++++++++----------------- 1 files changed, 53 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44..c94d4c1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; struct uda134x_platform_data *pd = codec->control_data; int i; u8 *cache = codec->reg_cache; @@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* ADC, DAC on */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg | 0x03); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - /* ADC, DAC power off */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +/* UDA1341 has the DAC/ADC power down in STATUS1 */ +static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), +}; + +/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ +static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->card->dev->platform_data; + const struct snd_soc_dapm_widget *widgets; + unsigned num_widgets; int ret; @@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) else uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pd->model == UDA134X_UDA1341) { + widgets = uda1341_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); + } else { + widgets = uda1340_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); + } + + ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + if (ret) { + printk(KERN_ERR "%s failed to register dapm controls: %d", + __func__, ret); + kfree(uda134x); + return ret; + } + switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: @@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .read = uda134x_read_reg_cache, .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .dapm_widgets = uda134x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), + .dapm_routes = uda134x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), }; static int uda134x_codec_probe(struct platform_device *pdev) -- 1.7.4.4

2 1

[alsa-devel] [PATCH 1/3] ALSA: hda - Clearing jackpoll_interval avoid pending work
by Wang Xingchao 30 Jul '13

30 Jul '13

Clearing jackpoll_interval before calling cancel_delayed_work_sync(), otherwise the work will be triggered again and cause impact in hda_jackpoll_work(). The next patch will poll jack once even with jackpoll_interval=0. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e2481ba..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } -- 1.8.3.2

1 2

[alsa-devel] [PATCH - microII 1/1] ALSA: usb-audio: Add Audio Advantage Micro II
by Przemek Rudy 30 Jul '13

30 Jul '13

Signed-off-by: Przemek Rudy <prudy1(a)o2.pl> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..d42a584 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -9,6 +9,8 @@ * Alan Cox (alan(a)lxorguk.ukuu.org.uk) * Thomas Sailer (sailer(a)ife.ee.ethz.ch) * + * Audio Advantage Micro II support added by: + * Przemek Rudy (prudy1(a)o2.pl) * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -30,6 +32,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/control.h> #include <sound/hwdep.h> @@ -1315,6 +1318,211 @@ static struct std_mono_table ebox44_table[] = { {} }; +/* Audio Advantage Micro II findings: + * + * Mapping spdif AES bits to vendor register.bit: + * AES0: [0 0 0 0 2.3 2.2 2.1 2.0] - default 0x00 + * AES1: [3.3 3.2.3.1.3.0 2.7 2.6 2.5 2.4] - default: 0x01 + * AES2: [0 0 0 0 0 0 0 0] + * AES3: [0 0 0 0 0 0 x 0] - 'x' bit is set basing on standard usb request + * (UAC_EP_CS_ATTR_SAMPLE_RATE) for Audio Devices + * + * power on values: + * r2: 0x10 + * r3: 0x20 (b7 is zeroed just before playback (except IEC61937) and set + * just after it to 0xa0, presumably it disables/mutes some analog + * parts when there is no audio.) + * r9: 0x28 + * + * Optical transmitter on/off: + * vendor register.bit: 9.1 + * 0 - on (0x28 register value) + * 1 - off (0x2a register value) + * + */ +static int snd_microii_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned int ep; + unsigned char data[3]; + int rate; + + ucontrol->value.iec958.status[0] = kcontrol->private_value & 0xff; + ucontrol->value.iec958.status[1] = (kcontrol->private_value >> 8) & 0xff; + ucontrol->value.iec958.status[2] = 0x00; + + /* use known values for that card: interface#1 altsetting#1 */ + iface = usb_ifnum_to_if(mixer->chip->dev, 1); + alts = &iface->altsetting[1]; + ep = get_endpoint(alts, 0)->bEndpointAddress; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + ep, + data, + sizeof(data)); + if (err < 0) + goto end; + + rate = data[0] | (data[1] << 8) | (data[2] << 16); + ucontrol->value.iec958.status[3] = (rate == 48000) ? + IEC958_AES3_CON_FS_48000 : IEC958_AES3_CON_FS_44100; + + err = 0; +end: + return err; +} + +static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg; + unsigned long priv_backup = kcontrol->private_value; + + reg = ((ucontrol->value.iec958.status[1] & 0x0f) << 4) | + (ucontrol->value.iec958.status[0] & 0x0f); + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 2, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xfffff0f0; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; + kcontrol->private_value |= (ucontrol->value.iec958.status[0] & 0x0f); + + reg = (ucontrol->value.iec958.status[0] & IEC958_AES0_NONAUDIO) ? + 0xa0 : 0x20; + reg |= (ucontrol->value.iec958.status[1] >> 4) & 0x0f; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 3, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xffff0fff; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; + + /* The frequency bits in AES3 cannot be set via register access. */ + + /* Silently ignore any bits from the request that cannot be set. */ + + err = (priv_backup != kcontrol->private_value); +end: + return err; +} + +static int snd_microii_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0x0f; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0x00; + ucontrol->value.iec958.status[3] = 0x00; + + return 0; +} + +static int snd_microii_spdif_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = !(kcontrol->private_value & 0x02); + + return 0; +} + +static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 9, + NULL, + 0); + + if (!err) { + err = (reg != (kcontrol->private_value & 0x0ff)); + if (err) + kcontrol->private_value = reg; + } + + return err; +} + +static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_default_get, + .put = snd_microii_spdif_default_put, + .private_value = 0x00000100UL,/* reset value */ + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = snd_microii_spdif_switch_get, + .put = snd_microii_spdif_switch_put, + .private_value = 0x00000028UL,/* reset value */ + } +}; + +static int snd_microii_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_microii_mixer_spdif); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_microii_mixer_spdif[i], mixer)); + if (err < 0) + return err; + } + + return err; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1353,6 +1561,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; + case USB_ID(0x0d8c, 0x0103): /* Audio Advantage Micro II */ + err = snd_microii_controls_create(mixer); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8b75bcf..2be7410 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3434,4 +3434,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ + USB_DEVICE(0x0d8c, 0x0103), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .product_name = "Audio Advantage MicroII", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 1.8.1.4

3 8

[alsa-devel] [PATCH - Preliminary support for Xonar Xense (resubmitAdded support for Xonar Xense. Only speaker out and mic in working.
by Ian Dawes 29 Jul '13

29 Jul '13

Signed-off-by: Ian Dawes <madeallup.gen(a)gmail.com> diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..df99ec9 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c..2567810 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -499,6 +499,17 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -1138,6 +1149,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + break; default: return -EINVAL; } -- 1.8.1.2

2 1

Re: [alsa-devel] snd_seq_system_info_get_cur_ports function
by Jose B 29 Jul '13

29 Jul '13

Dear Devs, I've been working on a wrapper to oversimplify using the alsa midi sequencer, and for it I was writing a function to quickly check if any new ports were registered on the sequencer after the application using the wrapper starts up. I noticed there is no function for getting the current port count but there is one for getting the current number of sequencer clients : snd_seq_system_info_get_cur_clients(const snd_seq_system_info_t *info) Shouldnt there be a corresponding function for getting the current port count as seen below? Or is it just missing from the documentation? snd_seq_system_info_get_cur_ports(const snd_seq_system_info_t *info) -- It doesn't show in code::blocks' autocomplete so i'm guessing it wasn't written yet. Is anyone willing to check this out please? It would be really handy to have this func. Thanks, and happy dev-ing Jose'

2 1

[alsa-devel] [PATCH] ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring
by Mark Brown 29 Jul '13

29 Jul '13

From: Mark Brown <broonie(a)linaro.org> As part of the multiplatform refactoring for AC'97 the AC'97 bus ops were staticised meaning that the prototype (which was never needed) conflicts with the declaration causing build failures. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/blackfin/bf5xx-ac97.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e..0c3e22d 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,7 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97_bus_ops bf5xx_ac97_ops; extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/2] tea575x: Move from sound to media
by Ondrej Zary 29 Jul '13

29 Jul '13

Hello, TEA575x is neither a sound device nor an i2c device. Let's finally move it from sound/i2c/other to drivers/media/radio. Tested with snd-es1968, snd-fm801 and radio-sf16fmr2. -- Ondrej Zary

3 5

[alsa-devel] [PATCH 1/2] ASoC: core: Add snd_soc_card_get_kcontrol()
by Dimitris Papastamos 29 Jul '13

29 Jul '13

This is useful for drivers who want to grab a pointer to snd_kcontrol outside of the kcontrol callbacks. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7..b33d1de 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -475,6 +475,8 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, const char *long_name, const char *prefix); +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d82ee38..d68a486 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2299,6 +2299,22 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, return 0; } +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + if (unlikely(!name)) + return NULL; + + list_for_each_entry(kctl, &card->controls, list) + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + return kctl; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); + /** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were -- 1.8.3.4

2 2

[alsa-devel] [PATCH] ALSA: hda - Remove analog mic pin override from STAC9228 dell-bios quirk
by Takashi Iwai 29 Jul '13

29 Jul '13

The current fixup for dell-bios model with STAC9228 codec contains the override of pin 0x0c for analog mic. But this is actually just adding a bogus pin and confuses the parser. Better to remove it for the auto-mic switching. Meanwhile, for a possible regression, keep the old configuration as model=dell-bios-amic, so that people can test it again quickly. Tested on Dell 1420n laptop. Reported-and-tested-by: Eric Shattow <lucent(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++-- 2 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 809d72b..a46ddb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -244,6 +244,7 @@ STAC9227/9228/9229/927x 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + dell-bios-amic Fixes with Dell BIOS setup including analog mic volknob Fixes with volume-knob widget 0x24 auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f8359..8f6c357 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_VERBS, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_DELL_BIOS_AMIC, STAC_DELL_BIOS_SPDIF, STAC_927X_DELL_DMIC, STAC_927X_VOLKNOB, @@ -3228,8 +3229,6 @@ static const struct hda_fixup stac927x_fixups[] = { [STAC_DELL_BIOS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - /* configure the analog microphone on some laptops */ - { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ { 0x0f, 0x0227011f }, /* correct the front input jack as a mic */ @@ -3239,6 +3238,16 @@ static const struct hda_fixup stac927x_fixups[] = { .chained = true, .chain_id = STAC_927X_DELL_DMIC, }, + [STAC_DELL_BIOS_AMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* configure the analog microphone on some laptops */ + { 0x0c, 0x90a79130 }, + {} + }, + .chained = true, + .chain_id = STAC_DELL_BIOS, + }, [STAC_DELL_BIOS_SPDIF] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3267,6 +3276,7 @@ static const struct hda_model_fixup stac927x_models[] = { { .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" }, { .id = STAC_DELL_3ST, .name = "dell-3stack" }, { .id = STAC_DELL_BIOS, .name = "dell-bios" }, + { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" }, { .id = STAC_927X_VOLKNOB, .name = "volknob" }, {} }; -- 1.8.3.1

1 0

[alsa-devel] [patch] ALSA: usx2y: remove an unneeded check
by Dan Carpenter 29 Jul '13

29 Jul '13

The test here is always true because S[i].urb is an array not a pointer. Also it's bogus because the intent was to test: if (S->urb[i]) { instead of: if (S[i].urb) { Anway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can just remove this. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 1f9bbd5..5a51b18 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S) { int i; for (i = 0; i < URBS_AsyncSeq; ++i) { - if (S[i].urb) { - usb_kill_urb(S->urb[i]); - usb_free_urb(S->urb[i]); - S->urb[i] = NULL; - } + usb_kill_urb(S->urb[i]); + usb_free_urb(S->urb[i]); + S->urb[i] = NULL; } kfree(S->buffer); }

2 1

[alsa-devel] [PATCH] ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION
by Vinod Koul 29 Jul '13

29 Jul '13

the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the return value was never updated for this call assign return value from put_user() Reported-by: Haynes <hgeorge(a)codeaurora.org> CC: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- 1.7.0.4

2 1

[alsa-devel] [PATCH - Preliminary Xonar Xense support 1/1] Added support for Xonar Xense. Only speaker out and mic in working.
by Gen 29 Jul '13

29 Jul '13

Signed-off-by: Gen <madeallup.gen(a)gmail.com> diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..df99ec9 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c..2567810 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -499,6 +499,17 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -1138,6 +1149,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + break; default: return -EINVAL; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Selectively call snd_hda_shutup_pins()
by Takashi Iwai 29 Jul '13

29 Jul '13

Instead of calling snd_hda_shutup_pins() unconditionally, allow it be called in spec->shutup callback. In this way, we can avoid calling this function if it causes a problem like we see in the next patch following this. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..dbd59df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); msleep(200); + snd_hda_shutup_pins(codec); } /* generic EAPD initialization */ @@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); - snd_hda_shutup_pins(codec); + else + snd_hda_shutup_pins(codec); } #define alc_free snd_hda_gen_free @@ -2573,15 +2575,13 @@ static void alc269_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return; - if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } + snd_hda_shutup_pins(codec); } static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION
by Vinod Koul 29 Jul '13

29 Jul '13

Reported-by: Haynes <hgeorge(a)codeaurora.org> CC: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- 1.7.0.4

2 2

[alsa-devel] Broken headphone and speaker amplifier supplies for the rt5640 driver
by Lars-Peter Clausen 29 Jul '13

29 Jul '13

Hi, The rt5640 driver sets for both the headphone and speaker amplifier widget the shift to SND_SOC_NOPM, which is -1. This doesn't make much sense and breaks compilation with some changes I've made. I guess for the heaphone amplifier the right fix is to set the shift to RT5640_PWR_HA_BIT. For the speaker amplifier I'd assume that the right bit is RT5640_PWR_CLS_D_BIT. But the speaker amplifier widget also as an event callback in which it already seems to set/clear the bit. I think the right fix is to remove the register change from the event callback and set the shift to RT5640_PWR_CLS_D_BIT. But I neither have the hardware nor the datasheet, so this is all just an educated guess. - Lars

4 6

Re: [alsa-devel] Broken headphone and speaker amplifier suppliesforthe rt5640 driver
by Bard Liao 29 Jul '13

29 Jul '13

> -----Original Message----- > From: alsa-devel-bounces(a)alsa-project.org > [mailto:alsa-devel-bounces@alsa-project.org] On Behalf Of Stephen Warren > Sent: Saturday, July 27, 2013 12:58 AM > To: Lars-Peter Clausen > Cc: Bard Liao; Linux-ALSA > Subject: Re: [alsa-devel] Broken headphone and speaker amplifier suppliesfor > the rt5640 driver > > On 07/26/2013 09:06 AM, Lars-Peter Clausen wrote: > > Hi, > > > > The rt5640 driver sets for both the headphone and speaker amplifier > > widget the shift to SND_SOC_NOPM, which is -1. This doesn't make much > > sense and breaks compilation with some changes I've made. > > > > I guess for the heaphone amplifier the right fix is to set the shift > > to RT5640_PWR_HA_BIT ... > > If I make those changes (see below for the exact patch I tried), then both > headphones and speakers do appear to still work fine. Bard will have to > comment on whether the changes are actually correct though:-) The control bits for headphone amplifier and speaker amplifier are both correct. But rt5640 need a sequence for headphone depop. So, I would like to do something like this. - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), I will send a patch ASAP. Thanks. > > > diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c > > index ce585e3..e3ab0b2 100644 > > --- a/sound/soc/codecs/rt5640.c > > +++ b/sound/soc/codecs/rt5640.c > > @@ -899,8 +899,6 @@ static int spk_event(struct snd_soc_dapm_widget > > *w, > > > > switch (event) { > > case SND_SOC_DAPM_POST_PMU: > > - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, > > - 0x0001, 0x0001); > > regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, > > 0xf000, 0xf000); > > break; > > @@ -908,8 +906,6 @@ static int spk_event(struct snd_soc_dapm_widget > *w, > > case SND_SOC_DAPM_PRE_PMD: > > regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, > > 0xf000, 0x0000); > > - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, > > - 0x0001, 0x0000); > > break; > > > > default: > > @@ -1159,13 +1155,13 @@ static const struct snd_soc_dapm_widget > rt5640_dapm_widgets[] = { > > SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", > RT5640_PWR_ANLG1, > > RT5640_PWR_MA_BIT, 0, NULL, 0), > > SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, > > - SND_SOC_NOPM, 0, NULL, 0), > > + RT5640_PWR_HA_BIT, 0, NULL, 0), > > SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, > > RT5640_PWR_HP_L_BIT, 0, NULL, 0), > > SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, > > RT5640_PWR_HP_R_BIT, 0, NULL, 0), > > SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, > > - SND_SOC_NOPM, 0, spk_event, > > + RT5640_PWR_CLS_D_BIT, 0, spk_event, > > SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), > > /* Output Lines */ > > SND_SOC_DAPM_OUTPUT("SPOLP"), > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH 0/6] R-Car sound driver support
by Kuninori Morimoto 29 Jul '13

29 Jul '13

Hi Mark These are basic Renesas R-Car sound driver support. Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2). This driver aims at supporting both Gen1/Gen2. The basic driver style must be satisfactory. But it was tested on Gen1 board only at this point. Gen2 will be supported soon. And, it supports PIO transfer only at this point, DMA transfer will be supported soon. These are based on mark/for-next branch Kuninori Morimoto (6): ASoC: add Renesas R-Car core feature ASoC: add Renesas R-Car module feature ASoC: add Renesas R-Car Generation feature ASoC: add Renesas R-Car SCU feature ASoC: add Renesas R-Car ADG feature ASoC: add Renesas R-Car SSI feature include/sound/rcar_snd.h | 75 +++++ sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 +++++++++++++++ sound/soc/sh/rcar/core.c | 699 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 306 +++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 253 ++++++++++++++++ sound/soc/sh/rcar/scu.c | 124 ++++++++ sound/soc/sh/rcar/ssi.c | 581 ++++++++++++++++++++++++++++++++++++ 10 files changed, 2284 insertions(+) Best regards --- Kuninori Morimoto

2 35

[alsa-devel] [PATCH 1/2] ASoC: lm4857: Use table based setup for DAPM and controls
by Lars-Peter Clausen 28 Jul '13

28 Jul '13

Let the ASoC core take care of registering the DAPM widget and routes as well as the controls. This makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/lm4857.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f595..5ea2ed0 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -174,28 +174,9 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { static int lm4857_probe(struct snd_soc_codec *codec) { struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; codec->control_data = lm4857->i2c; - ret = snd_soc_add_codec_controls(codec, lm4857_controls, - ARRAY_SIZE(lm4857_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, - ARRAY_SIZE(lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, - ARRAY_SIZE(lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_new_widgets(dapm); - return 0; } @@ -207,6 +188,13 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .reg_word_size = sizeof(uint8_t), .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, + + .controls = lm4857_controls, + .num_controls = ARRAY_SIZE(lm4857_controls), + .dapm_widgets = lm4857_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets), + .dapm_routes = lm4857_routes, + .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; static int lm4857_i2c_probe(struct i2c_client *i2c, -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: compress: use soc_xxx handlers for metadata
by Vinod Koul 28 Jul '13

28 Jul '13

the compress metadata handlers were wrongly named sst_xxx Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-compress.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 06a8000..d220150 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -334,7 +334,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_set_metadata(struct snd_compr_stream *cstream, +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -347,7 +347,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_get_metadata(struct snd_compr_stream *cstream, +static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -364,8 +364,8 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, - .set_metadata = sst_compr_set_metadata, - .get_metadata = sst_compr_get_metadata, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

2 1

[alsa-devel] RFC: support for 12 & 24Khz
by Vinod Koul 28 Jul '13

28 Jul '13

Hey Takashi, For compressed audio we also need to support the PCM rates of 12 and 24KHz. Looking at pcm.h these are not defined so we can simply add them at the end. But am worried about wider impact of adding these rates. Can you let me know if more is required to be done or below is fine diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 84b10f9..e418d8d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -126,6 +126,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_12000 <1<<13> /* 12000Hz */ +#define SNDRV_PCM_RATE_24000 <1<<14> /* 24000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..42600b0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1779,12 +1779,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_24000 != 1 << 14 #error "Change this table" #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000, + 12000, 24000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), ~Vinod --

5 11

[alsa-devel] Garbage at end of playback: ecasound or alsa broken?
by Knut Petersen 28 Jul '13

28 Jul '13

Hi everybody! ====================================================================== Summary: A very short period (about 45ms) garbage is played at the end of every playback. ====================================================================== Hardware ======== mobo: AOpen i915GMm-hfs cpu: Pentium-M ram: 2GB intel hda, realtek codec rme digi96/pad onboard intel hda, realtek alc880 Software ======= opensuse 12.3 linux kernel 3.10.1, 3.11-rc1 ecasound, git ab6a9c3be5440b05f2c9eb240ee0d42578379f8d Steps to reproduce the problem: ========================= 1: generate test file =============== Start audacity, generate a 44.1 khz stereo track, one second audio, start 400 Hz, end 4000Hz, start amplitude 0.5, end amplitude 1.0 export to testin.wav as 16bit wav 2: playback test file and record it ========================= Setup soundcards and cabling, execute "ecasound -t 2 -f:16,2,44100 -a:1 -i testin.wav -o alsahw,0,0 -a:2 -i alsahw,0,0 -o testout.wav" 3:Expected result: ============== testout.wav should hold a proper recording of testin.wav, padded with silence at the end. 4: my result ========= testout.wav 1 - starts with about 90 Samples of silence (not nice, how can I avoid that?) 2 - continues with 1 second of properly recorded audio (expected) 3 - continues with about 1982 samples of audio (0,045s) (unexpected, severly broken) The garbage audio is part of the original audio, starting at about 26xxHz, ending at about 28xx Hz 4 - continues with silence to the end of the file (expected) Discussion ========= The same problem exists if I record rme96 to rme96, hda to rme96 or hda to hda. So it´s probably not a problem of the rme96 or hda alsa drivers. Is this problem known? Can anybody confirm the problem with different hardware? Should I blame alsa or ecasound? Any ideas? cu, Knut

2 1

[alsa-devel] [PATCH 1/3] ASoC: smdk_wm8994: Make driver name more unique
by Mark Brown 27 Jul '13

27 Jul '13

From: Mark Brown <broonie(a)linaro.org> Avoid collisions with other SMDK audio by using the CODEC name. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/smdk_wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 581ea4a..05c479c 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -200,7 +200,7 @@ MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio", + .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, @@ -212,4 +212,4 @@ module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:smdk-audio"); +MODULE_ALIAS("platform:smdk-audio-wm8994"); -- 1.8.3.2

1 2

Re: [alsa-devel] [PATCH 35/50] media: usb: cx231xx: spin_lock in complete() cleanup
by Hans Verkuil 27 Jul '13

27 Jul '13

On 07/11/2013 11:05 AM, Ming Lei wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: Hans Verkuil <hans.verkuil(a)cisco.com> > Cc: linux-media(a)vger.kernel.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > drivers/media/usb/cx231xx/cx231xx-audio.c | 6 ++++++ > drivers/media/usb/cx231xx/cx231xx-core.c | 10 ++++++---- > drivers/media/usb/cx231xx/cx231xx-vbi.c | 5 +++-- > 3 files changed, 15 insertions(+), 6 deletions(-) > > diff --git a/drivers/media/usb/cx231xx/cx231xx-audio.c b/drivers/media/usb/cx231xx/cx231xx-audio.c > index 81a1d97..58c1b5c 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-audio.c > +++ b/drivers/media/usb/cx231xx/cx231xx-audio.c > @@ -136,6 +136,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > stride = runtime->frame_bits >> 3; > > for (i = 0; i < urb->number_of_packets; i++) { > + unsigned long flags; > int length = urb->iso_frame_desc[i].actual_length / > stride; > cp = (unsigned char *)urb->transfer_buffer + > @@ -158,6 +159,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > length * stride); > } > > + local_irq_save(flags); > snd_pcm_stream_lock(substream); Can't you use snd_pcm_stream_lock_irqsave here? Ditto for the other media drivers where this happens: em28xx and tlg2300. I've reviewed the media driver changes and they look OK to me, so if my comment above is fixed, then I can merge them for 3.12. Or are these changes required for 3.11? Regards, Hans > > dev->adev.hwptr_done_capture += length; > @@ -174,6 +176,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > period_elapsed = 1; > } > snd_pcm_stream_unlock(substream); > + local_irq_restore(flags); > } > if (period_elapsed) > snd_pcm_period_elapsed(substream); > @@ -224,6 +227,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > stride = runtime->frame_bits >> 3; > > if (1) { > + unsigned long flags; > int length = urb->actual_length / > stride; > cp = (unsigned char *)urb->transfer_buffer; > @@ -242,6 +246,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > length * stride); > } > > + local_irq_save(flags); > snd_pcm_stream_lock(substream); > > dev->adev.hwptr_done_capture += length; > @@ -258,6 +263,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > period_elapsed = 1; > } > snd_pcm_stream_unlock(substream); > + local_irq_restore(flags); > } > if (period_elapsed) > snd_pcm_period_elapsed(substream); > diff --git a/drivers/media/usb/cx231xx/cx231xx-core.c b/drivers/media/usb/cx231xx/cx231xx-core.c > index 4ba3ce0..593b397 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-core.c > +++ b/drivers/media/usb/cx231xx/cx231xx-core.c > @@ -798,6 +798,7 @@ static void cx231xx_isoc_irq_callback(struct urb *urb) > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, video_mode); > int i; > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -813,9 +814,9 @@ static void cx231xx_isoc_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->video_mode.slock); > + spin_lock_irqsave(&dev->video_mode.slock, flags); > dev->video_mode.isoc_ctl.isoc_copy(dev, urb); > - spin_unlock(&dev->video_mode.slock); > + spin_unlock_irqrestore(&dev->video_mode.slock, flags); > > /* Reset urb buffers */ > for (i = 0; i < urb->number_of_packets; i++) { > @@ -842,6 +843,7 @@ static void cx231xx_bulk_irq_callback(struct urb *urb) > struct cx231xx_video_mode *vmode = > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, video_mode); > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -857,9 +859,9 @@ static void cx231xx_bulk_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->video_mode.slock); > + spin_lock_irqsave(&dev->video_mode.slock, flags); > dev->video_mode.bulk_ctl.bulk_copy(dev, urb); > - spin_unlock(&dev->video_mode.slock); > + spin_unlock_irqrestore(&dev->video_mode.slock, flags); > > /* Reset urb buffers */ > urb->status = usb_submit_urb(urb, GFP_ATOMIC); > diff --git a/drivers/media/usb/cx231xx/cx231xx-vbi.c b/drivers/media/usb/cx231xx/cx231xx-vbi.c > index c027942..38e78f8 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-vbi.c > +++ b/drivers/media/usb/cx231xx/cx231xx-vbi.c > @@ -306,6 +306,7 @@ static void cx231xx_irq_vbi_callback(struct urb *urb) > struct cx231xx_video_mode *vmode = > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, vbi_mode); > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -322,9 +323,9 @@ static void cx231xx_irq_vbi_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->vbi_mode.slock); > + spin_lock_irqsave(&dev->vbi_mode.slock, flags); > dev->vbi_mode.bulk_ctl.bulk_copy(dev, urb); > - spin_unlock(&dev->vbi_mode.slock); > + spin_unlock_irqrestore(&dev->vbi_mode.slock, flags); > > /* Reset status */ > urb->status = 0; >

2 1

[alsa-devel] [PATCH] ASoc: kirkwood: free the external clock when same as the internal one
by Jean-Francois Moine 27 Jul '13

27 Jul '13

Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/{kirkwood-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ba72039..bbadf53 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c~ +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -495,6 +495,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { + devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

Re: [alsa-devel] Decibel Levels for Storm Medusa 5.1 USB Headset
by Raymond Yau 27 Jul '13

27 Jul '13

> >>> I hope this Information helps. I have the Storm Medusa 5.1 USB Headset > >>> which reports wrong decibel levels to pulseaudio. My Alsa-Info are > >>> located at > >>> > >>> > > http://www.alsa-project.org/db/?f=a882f3db6769c5dfce83c266946eff52d1d55349 > >>> The Headset is card2 in the above dump. > >>> > >>> My decibel levels-fix > >>> > >>> [DecibelFix Speaker] > >>> db-values = 1:-30 37:3 > >> How did you define these values? > > http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/003867.h… > > > > If your headset have windows driver, need to find out which USB chip and > > look at the datasheet of the chip > > > > e.g. cm106l ’s datasheet contain a table of dac volume cortol > Ok looks like cm6206 is only in newer versions of the headset in older > (like mine) there is cm106l , maybe there are other chips as well but i > don't know. http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/003877.h… Seem same USB audio id idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0006 Do you find out the audio chip from product spec, user manual or windows driver ? Master volume control by default; per-channel volume control by C-Media driver Refer to the function block diagram , dac with volume controls 0 to -45dB There is a table of dac volume control in linear scale

1 0

[alsa-devel] [PATCH 0/9] ARM: tegra: move to generic DMA DT binding
by Richard Zhao 26 Jul '13

26 Jul '13

The patch series aim to move apbdma to generic DT binding. Changes: - It add dmas/dma-names properties while leave nvidia,dma-request-selector to be compatible. - update apbdma driver and dma client drivers - ASoC tegra adds a new struct to pass device and dma-name. - at last, remove legacy nvidia,dma-request-selector Richard Zhao (9): ARM: dts: add generic DMA DT binding for tegra apbdma dma: tegra20-apbdma: move to generic device tree bindings spi: tegra114: move to generic dma DT binding spi: tegra20-slink: move to generic dma DT binding spi: tegra20-sflash: move to generic dma DT binding serial: tegra: move to generic dma DT binding ASoC: tegra: move to generic DMA DT binding ARM: dts: tegra: remove legacy nvidia,dma-request-selector properties dma: tegra20-apbdma: remove legacy nvidia,dma-request-selector support .../devicetree/bindings/dma/tegra20-apbdma.txt | 1 + .../bindings/serial/nvidia,tegra20-hsuart.txt | 8 ++-- .../bindings/sound/nvidia,tegra20-ac97.txt | 8 ++-- .../bindings/sound/nvidia,tegra20-i2s.txt | 8 ++-- .../bindings/sound/nvidia,tegra30-ahub.txt | 12 +++--- .../bindings/spi/nvidia,tegra114-spi.txt | 10 +++-- .../bindings/spi/nvidia,tegra20-sflash.txt | 10 +++-- .../bindings/spi/nvidia,tegra20-slink.txt | 10 +++-- arch/arm/boot/dts/tegra114.dtsi | 41 +++++++++++++------- arch/arm/boot/dts/tegra20.dtsi | 40 +++++++++++++------- arch/arm/boot/dts/tegra30.dtsi | 37 ++++++++++++------ drivers/dma/tegra20-apb-dma.c | 38 ++++++++++++++++++- drivers/spi/spi-tegra114.c | 16 +++----- drivers/spi/spi-tegra20-slink.c | 16 +++----- drivers/tty/serial/serial-tegra.c | 16 +------- sound/soc/tegra/tegra20_ac97.c | 17 +++------ sound/soc/tegra/tegra20_ac97.h | 2 + sound/soc/tegra/tegra20_i2s.c | 26 ++++--------- sound/soc/tegra/tegra20_i2s.h | 2 + sound/soc/tegra/tegra30_ahub.c | 25 +++++------- sound/soc/tegra/tegra30_ahub.h | 11 +++--- sound/soc/tegra/tegra30_i2s.c | 44 +++++++++++++++------- sound/soc/tegra/tegra30_i2s.h | 2 + sound/soc/tegra/tegra_pcm.c | 14 +++++++ sound/soc/tegra/tegra_pcm.h | 13 +++++++ 25 files changed, 260 insertions(+), 167 deletions(-) -- 1.8.1.5

3 12

[alsa-devel] [patch] ASoC: adau1701: type bug with ADAU1707_CLKDIV_UNSET
by Dan Carpenter 26 Jul '13

26 Jul '13

ADAU1707_CLKDIV_UNSET is always compared against an unsigned int and not an unsigned long. The current tests are always false. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 44d8a95..2c10252 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -91,7 +91,7 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 -#define ADAU1707_CLKDIV_UNSET (-1UL) +#define ADAU1707_CLKDIV_UNSET (-1U) #define ADAU1701_FIRMWARE "adau1701.bin"

3 2

[alsa-devel] [PATCH] ASoC: wm8994: Add clock bindings to the device tree
by Mark Brown 26 Jul '13

26 Jul '13

From: Mark Brown <broonie(a)linaro.org> Due to the variable availability of the clock API there is no code in the driver to use these at present, for the time being the machine drivers will need to handle requesting clocks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Documentation/devicetree/bindings/sound/wm8994.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index f2f3e80..e045e90 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -32,6 +32,10 @@ Optional properties: The second cell is the flags, encoded as the trigger masks from Documentation/devicetree/bindings/interrupts.txt + - clocks : A list of up to two phandle and clock specifier pairs + - clock-names : A list of clock names sorted in the same order as clocks. + Valid clock names are "MCLK1" and "MCLK2". + - wlf,gpio-cfg : A list of GPIO configuration register values. If absent, no configuration of these registers is performed. If any value is over 0xffff then the register will be left as default. If present 11 -- 1.8.3.2

3 5

[alsa-devel] [RFC PATCH 0/2] tea575x: Move from sound to media
by Ondrej Zary 26 Jul '13

26 Jul '13

Hello, TEA575x is neither a sound device nor an i2c device. Let's finally move it from sound/i2c/other to drivers/media/radio. Tested with snd-es1968, snd-fm801 and radio-sf16fmr2. I guess the Kconfig dependencies are not correct. -- Ondrej Zary

2 1

[alsa-devel] [PATCH 4/4] ASoc: kirkwood: add DT support
by Jean-Francois Moine 26 Jul '13

26 Jul '13

The kirkwood audio driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/kirkwood-i2s.txt | 29 ++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 34 +++++++++++++++------- 2 files changed, 52 insertions(+), 11 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/kirkwood-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt new file mode 100644 index 0000000..015a769 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) I2S controller + +Required properties: + +- compatible: "marvell,mvebu-i2s" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "external" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-i2s"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3517784..fbe5150 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -12,7 +12,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/io.h> #include <linux/slab.h> #include <linux/mbus.h> @@ -22,6 +21,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "kirkwood-i2s" @@ -497,16 +498,18 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); - return -EINVAL; - } - - priv->burst = data->burst; + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); + return -EINVAL; + } - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, "internal"); if (IS_ERR(priv->clk)) { - dev_err(&pdev->dev, "no clock\n"); + dev_err(&pdev->dev, "no internal clock\n"); return PTR_ERR(priv->clk); } @@ -514,7 +517,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (err < 0) return err; - priv->extclk = clk_get(&pdev->dev, "extclk"); + priv->extclk = clk_get(&pdev->dev, "external"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { clk_put(priv->extclk); @@ -531,7 +534,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -556,12 +559,21 @@ fail: return err; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "marvell,mvebu-i2s" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 6

[alsa-devel] [PATCH 3/4] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Jean-Francois Moine 26 Jul '13

26 Jul '13

To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, this patch merges the kirkwood-i2c and kirkwood-dma drivers into one module. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- arch/arm/mach-kirkwood/common.c | 6 ------ sound/soc/kirkwood/Kconfig | 5 ----- sound/soc/kirkwood/Makefile | 4 +--- sound/soc/kirkwood/kirkwood-dma.c | 30 +----------------------------- sound/soc/kirkwood/kirkwood-i2s.c | 5 +++++ sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 1 + 8 files changed, 10 insertions(+), 45 deletions(-) diff --git a/arch/arm/mach-kirkwood/common.c b/arch/arm/mach-kirkwood/common.c index e9238b5..fd02f1b 100644 --- a/arch/arm/mach-kirkwood/common.c +++ b/arch/arm/mach-kirkwood/common.c @@ -587,15 +587,9 @@ static struct platform_device kirkwood_i2s_device = { }, }; -static struct platform_device kirkwood_pcm_device = { - .name = "kirkwood-pcm-audio", - .id = -1, -}; - void __init kirkwood_audio_init(void) { platform_device_register(&kirkwood_i2s_device); - platform_device_register(&kirkwood_pcm_device); } /***************************************************************************** diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715..b6917a7 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -6,14 +6,10 @@ config SND_KIRKWOOD_SOC the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. -config SND_KIRKWOOD_SOC_I2S - tristate - config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) depends on I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help Say Y if you want to add support for SoC audio on @@ -22,7 +18,6 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help Say Y if you want to add support for SoC audio on diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 3e62ae9..cf22ad3 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,6 @@ -snd-soc-kirkwood-objs := kirkwood-dma.o -snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o +snd-soc-kirkwood-objs := kirkwood-i2s.o kirkwood-dma.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cef5277..9f4c595 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -364,36 +364,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver kirkwood_soc_platform = { +struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, }; - -static int kirkwood_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); -} - -static int kirkwood_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver kirkwood_pcm_driver = { - .driver = { - .name = "kirkwood-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = kirkwood_soc_platform_probe, - .remove = kirkwood_soc_platform_remove, -}; - -module_platform_driver(kirkwood_pcm_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-pcm-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d3e50af..3517784 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -459,6 +459,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) { @@ -544,6 +545,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto fail; } + err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);; + if (err) { + dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); + goto fail; return 0; fail: diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..0fa3082 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -55,7 +55,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .name = "CS42L51", .stream_name = "CS42L51 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..f48ebda 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -71,7 +71,7 @@ static struct snd_soc_dai_link t5325_dai[] = { .name = "ALC5621", .stream_name = "ALC5621 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 4d92637..f3de119 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -133,4 +133,5 @@ struct kirkwood_dma_data { int burst; }; +extern struct snd_soc_platform_driver kirkwood_soc_platform; #endif -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 2/4] ASoc: kirkwood: simplify probe error
by Jean-Francois Moine 26 Jul '13

26 Jul '13

The function kirkwood_i2s_dev_remove() may be used when probe fails. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 44 +++++++++++++++++++-------------------- 1 file changed, 21 insertions(+), 23 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..d3e50af 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -455,6 +455,21 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { .name = DRV_NAME, }; +static int kirkwood_i2s_dev_remove(struct platform_device *pdev) +{ + struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + + if (!IS_ERR(priv->extclk)) { + clk_disable_unprepare(priv->extclk); + clk_put(priv->extclk); + } + clk_disable_unprepare(priv->clk); + + return 0; +} + static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; @@ -525,34 +540,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, soc_dai, 1); - if (!err) - return 0; - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); - - if (!IS_ERR(priv->extclk)) { - clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + goto fail; } - clk_disable_unprepare(priv->clk); + return 0; +fail: + kirkwood_i2s_dev_remove(pdev); return err; } -static int kirkwood_i2s_dev_remove(struct platform_device *pdev) -{ - struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); - - snd_soc_unregister_component(&pdev->dev); - - if (!IS_ERR(priv->extclk)) { - clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); - } - clk_disable_unprepare(priv->clk); - - return 0; -} - static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 1/4] ASoc: kirkwood: change the pcm/dma stream data pointer
by Jean-Francois Moine 26 Jul '13

26 Jul '13

In the kirkwood pcm/dma driver, the private stream related data pointer was hold in the platform drvdata. As this pointer may be used by other parts of the audio subsystem, this patch uses the substream runtime private data pointer instead. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-dma.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index a9f1453..cef5277 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -127,10 +127,9 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) int err; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_platform *platform = soc_runtime->platform; struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; struct kirkwood_dma_data *priv; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + struct kirkwood_dma_priv *prdata = runtime->private_data; const struct mbus_dram_target_info *dram; unsigned long addr; @@ -171,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) return -EBUSY; } - snd_soc_platform_set_drvdata(platform, prdata); + runtime->private_data = prdata; /* * Enable Error interrupts. We're only ack'ing them but @@ -199,8 +198,7 @@ static int kirkwood_dma_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct snd_soc_platform *platform = soc_runtime->platform; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + struct kirkwood_dma_priv *prdata = substream->runtime->private_data; struct kirkwood_dma_data *priv; priv = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -217,7 +215,7 @@ static int kirkwood_dma_close(struct snd_pcm_substream *substream) writel(0, priv->io + KIRKWOOD_ERR_MASK); free_irq(priv->irq, prdata); kfree(prdata); - snd_soc_platform_set_drvdata(platform, NULL); + substream->runtime->private_data = NULL; } return 0; -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 1/3] ALSA: hda - Clearing jackpoll_interval avoid pending work
by Wang Xingchao 26 Jul '13

26 Jul '13

Clearing jackpoll_interval before calling cancel_delayed_work_sync(), otherwise the work will be triggered again and cause impact in hda_jackpoll_work(). The next patch will poll jack once even with jackpoll_interval=0. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e2481ba..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } -- 1.8.3.2

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[alsa-devel] Decibel Levels for Storm Medusa 5.1 USB Headset
by Alexander Blinne 26 Jul '13

26 Jul '13

Hello everyone, I hope this Information helps. I have the Storm Medusa 5.1 USB Headset which reports wrong decibel levels to pulseaudio. My Alsa-Info are located at http://www.alsa-project.org/db/?f=a882f3db6769c5dfce83c266946eff52d1d55349 The Headset is card2 in the above dump. My decibel levels-fix [DecibelFix Speaker] db-values = 1:-30 37:3 works fine. Level 0 is mute, hence -infinity dB. Greetings

3 3

[alsa-devel] [PATCH] ASoC: fsl: Set sdma peripheral type directly
by Nicolin Chen 25 Jul '13

25 Jul '13

Let CPU DAI drivers set SDMA periperal type directly to support more dma types(SPDIF, ESAI) other than only two for SSI. This will easily allow some non-SSI drivers to use the imx-pcm-dma as well. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- @Timur Compile-passed with mpc85xx_defconfig by using gcc-4.6.3-nolibc_powerpc-linux. --- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/imx-pcm.h | 7 ++----- sound/soc/fsl/imx-ssi.c | 4 ++-- 3 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 11469fe..4d78df7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -775,9 +775,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) "fsl,spba-bus"); imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx, - dma_events[0], shared); + dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, - dma_events[1], shared); + dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); } /* Initialize the the device_attribute structure */ diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index fd56cad..9136625 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,14 +22,11 @@ static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, - int dma, bool shared) + int dma, enum sdma_peripheral_type peripheral_type) { dma_data->dma_request = dma; dma_data->priority = DMA_PRIO_HIGH; - if (shared) - dma_data->peripheral_type = IMX_DMATYPE_SSI_SP; - else - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = peripheral_type; } struct imx_pcm_fiq_params { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f029e27..f58bcd8 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -571,13 +571,13 @@ static int imx_ssi_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start, - false); + IMX_DMATYPE_SSI); } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start, - false); + IMX_DMATYPE_SSI); } platform_set_drvdata(pdev, ssi); -- 1.7.1

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[alsa-devel] [PATCH 4/4 v2] ASoc: kirkwood: add DT support
by Jean-Francois Moine 25 Jul '13

25 Jul '13

The kirkwood audio driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 fix compilation error --- Documentation/devicetree/bindings/sound/kirkwood-i2s.txt | 29 ++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 35 +++++++++++++++------- 2 files changed, 53 insertions(+), 11 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/kirkwood-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt new file mode 100644 index 0000000..015a769 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) I2S controller + +Required properties: + +- compatible: "marvell,mvebu-i2s" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "external" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-i2s"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/home/jef/kernel/sound/soc/kirkwood/kirkwood-i2s.c.new index a1037e2..26c2f74 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/home/jef/kernel/sound/soc/kirkwood/kirkwood-i2s.c.new @@ -12,7 +12,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/io.h> #include <linux/slab.h> #include <linux/mbus.h> @@ -22,6 +21,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "kirkwood-i2s" @@ -477,6 +478,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -497,16 +499,18 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); - return -EINVAL; - } - - priv->burst = data->burst; + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); + return -EINVAL; + } - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, "internal"); if (IS_ERR(priv->clk)) { - dev_err(&pdev->dev, "no clock\n"); + dev_err(&pdev->dev, "no internal clock\n"); return PTR_ERR(priv->clk); } @@ -514,7 +518,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (err < 0) return err; - priv->extclk = clk_get(&pdev->dev, "extclk"); + priv->extclk = clk_get(&pdev->dev, "external"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { clk_put(priv->extclk); @@ -531,7 +535,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -557,12 +561,21 @@ fail: return err; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "marvell,mvebu-i2s" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 3/4 v2] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Jean-Francois Moine 25 Jul '13

25 Jul '13

To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, this patch merges the kirkwood-i2c and kirkwood-dma drivers into one module. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 - fix compilation error --- arch/arm/mach-kirkwood/common.c | 6 ------ sound/soc/kirkwood/Kconfig | 5 ----- sound/soc/kirkwood/Makefile | 4 +--- sound/soc/kirkwood/kirkwood-dma.c | 30 +----------------------------- sound/soc/kirkwood/kirkwood-i2s.c | 6 ++++++ sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 1 + 8 files changed, 11 insertions(+), 45 deletions(-) diff --git a/arch/arm/mach-kirkwood/common.c b/arch/arm/mach-kirkwood/common.c index e9238b5..fd02f1b 100644 --- a/arch/arm/mach-kirkwood/common.c +++ b/arch/arm/mach-kirkwood/common.c @@ -587,15 +587,9 @@ static struct platform_device kirkwood_i2s_device = { }, }; -static struct platform_device kirkwood_pcm_device = { - .name = "kirkwood-pcm-audio", - .id = -1, -}; - void __init kirkwood_audio_init(void) { platform_device_register(&kirkwood_i2s_device); - platform_device_register(&kirkwood_pcm_device); } /***************************************************************************** diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715..b6917a7 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -6,14 +6,10 @@ config SND_KIRKWOOD_SOC the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. -config SND_KIRKWOOD_SOC_I2S - tristate - config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) depends on I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help Say Y if you want to add support for SoC audio on @@ -22,7 +18,6 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help Say Y if you want to add support for SoC audio on diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 3e62ae9..cf22ad3 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,6 @@ -snd-soc-kirkwood-objs := kirkwood-dma.o -snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o +snd-soc-kirkwood-objs := kirkwood-i2s.o kirkwood-dma.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cef5277..9f4c595 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -364,36 +364,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver kirkwood_soc_platform = { +struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, }; - -static int kirkwood_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); -} - -static int kirkwood_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver kirkwood_pcm_driver = { - .driver = { - .name = "kirkwood-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = kirkwood_soc_platform_probe, - .remove = kirkwood_soc_platform_remove, -}; - -module_platform_driver(kirkwood_pcm_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-pcm-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d3e50af..3517784 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -459,6 +459,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) { @@ -544,6 +545,11 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto fail; } + err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);; + if (err) { + dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); + goto fail; + } return 0; fail: diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..0fa3082 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -55,7 +55,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .name = "CS42L51", .stream_name = "CS42L51 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..f48ebda 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -71,7 +71,7 @@ static struct snd_soc_dai_link t5325_dai[] = { .name = "ALC5621", .stream_name = "ALC5621 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 4d92637..f3de119 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -133,4 +133,5 @@ struct kirkwood_dma_data { int burst; }; +extern struct snd_soc_platform_driver kirkwood_soc_platform; #endif -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - jack poll once if jackpoll_interval==0
by Wang Xingchao 25 Jul '13

25 Jul '13

jackpoll_interval used to poll jack event periodically, if it's 0, give the caller one chance to probe jack status. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8a005f0..74c11bc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1216,11 +1216,12 @@ static void hda_jackpoll_work(struct work_struct *work) { struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - if (!codec->jackpoll_interval) - return; snd_hda_jack_set_dirty_all(codec); snd_hda_jack_poll_all(codec); + if (!codec->jackpoll_interval) + return; + queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, codec->jackpoll_interval); } -- 1.8.3.2

4 16

[alsa-devel] [PATCH 0/6] ASoC: add CSR SiRFSoC sound drivers
by Barry Song 25 Jul '13

25 Jul '13

This patchset adds CSR SiRFSoC sound drivers including: 1. the platform DMA driver which will be shard by all DAI 2. the I2S CPU DAI driver 3. the USP-based PCM CPU DAI driver 4. CPU DAI and Codec driver for internal chip codec 5. the mach driver for EVB board using internal codec Barry Song (1): arm: prima2: defconfig: enable sound components Rongjun Ying (5): ASoC: sirf: add sirf platform driver which provides DMA ASoC: sirf: add I2S CPU DAI driver ASoC: usp-pcm: add CPU DAI driver for PCM simulated from USP ASoC: sirf-soc-inner: add drivers for both CPU and Codec DAIs ASoC: sirf-inner: add mach driver for SiRFSoC internal codec arch/arm/configs/prima2_defconfig | 10 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sirf/Kconfig | 18 ++ sound/soc/sirf/Makefile | 11 + sound/soc/sirf/sirf-audio.h | 266 ++++++++++++++++ sound/soc/sirf/sirf-i2s.c | 411 ++++++++++++++++++++++++ sound/soc/sirf/sirf-inner.c | 267 ++++++++++++++++ sound/soc/sirf/sirf-pcm.c | 221 +++++++++++++ sound/soc/sirf/sirf-pcm.h | 17 + sound/soc/sirf/sirf-soc-inner.c | 653 ++++++++++++++++++++++++++++++++++++++ sound/soc/sirf/sirf-usp.c | 481 ++++++++++++++++++++++++++++ sound/soc/sirf/sirf-usp.h | 276 ++++++++++++++++ 13 files changed, 2630 insertions(+), 3 deletions(-) create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio.h create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-inner.c create mode 100644 sound/soc/sirf/sirf-pcm.c create mode 100644 sound/soc/sirf/sirf-pcm.h create mode 100644 sound/soc/sirf/sirf-soc-inner.c create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.8.2.3

3 18

[alsa-devel] [PATCH] ASoC: rt5640: add headset detection function
by bardliao＠realtek.com 25 Jul '13

25 Jul '13

From: Bard Liao <bardliao(a)realtek.com> Add headset detection function to rt5640 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 61 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 3 +++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..dc0d063 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1911,6 +1911,67 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define rt5640_resume NULL #endif +/** + * rt5640_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +int rt5640_headset_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int jack_insert) +{ + int jack_type = 0; + int sclk_src = 0; + int reg63, reg64; + + if (jack_insert) { + reg63 = snd_soc_read(codec, RT5640_PWR_ANLG1); + reg64 = snd_soc_read(codec, RT5640_PWR_ANLG2); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_write(codec, RT5640_PWR_ANLG1, 0xa814); + sclk_src = snd_soc_read(codec, RT5640_GLB_CLK) & + RT5640_SCLK_SRC_MASK; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, + 0x2 << RT5640_SCLK_SRC_SFT); + } + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LDO2, RT5640_PWR_LDO2); + snd_soc_update_bits(codec, RT5640_PWR_ANLG2, + RT5640_PWR_MB1, RT5640_PWR_MB1); + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK | RT5640_MIC1_OVTH_MASK | + RT5640_PWR_CLK25M_MASK | RT5640_PWR_MB_MASK, + RT5640_MIC1_OVCD_EN | RT5640_MIC1_OVTH_600UA | + RT5640_PWR_MB_PU | RT5640_PWR_CLK25M_PU); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0x1); + msleep(100); + if (snd_soc_read(codec, RT5640_IRQ_CTRL2) & 0x8) + jack_type = SND_JACK_HEADPHONE; + else + jack_type = SND_JACK_HEADSET; + snd_soc_update_bits(codec, RT5640_IRQ_CTRL2, + RT5640_MB1_OC_CLR, 0); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, sclk_src); + snd_soc_write(codec, RT5640_PWR_ANLG1, reg63); + snd_soc_write(codec, RT5640_PWR_ANLG2, reg64); + snd_soc_jack_report(jack, jack_type, SND_JACK_HEADSET); + } else { + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK, + RT5640_MIC1_OVCD_DIS); + snd_soc_jack_report(jack, 0, jack->jack->type); + } + + return jack_type; +} +EXPORT_SYMBOL_GPL(rt5640_headset_detect); + #define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..263615b 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2089,4 +2089,7 @@ struct rt5640_priv { int dmic_en; }; +int rt5640_headset_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int jack_insert); + #endif -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH 0/4] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 25 Jul '13

25 Jul '13

This patch series adds DT support to the kirkwood audio subsystem. It cancels the previous patch requests: - ARM: kirkwood: enable S/PDIF - ARM: kirkwood: extend the kirkwood i2s driver for DT usage - ARM: kirkwood: extend the kirkwood pcm/dma driver for DT usage It also contains the merge of kirkwood-i2c and kirkwood-dma into one module, and, the removal of the kirkwood-pcm-audio device. This involves changes in the openrd and t5325 drivers, as in the common Kirkwood machine initialization. These last changes have not been tested. -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] PCM extra attributes
by Vinod Koul 25 Jul '13

25 Jul '13

Hey Takashi, We were looking to find the substream index in the userspace for the device opened (in case we support multiple substreams and mix them). Since we also wanted to have some specific alsa controls for each substream the problem of not knowing substream in usermode make stuff interesting! I was pointed to an old RFC of yours [1] and it fits my needs. I can get the substream index using this and program the alsa controls. Now reading the archive looks like it was note merged. Would be okay to do so for above case. I can resurrect the patchset if you would like... Thanks ~Vinod [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018651.html --

3 3

[alsa-devel] [PATCH 0/4 V7] Power-well API implementation for Haswell
by Wang Xingchao 25 Jul '13

25 Jul '13

Hi all, This is V7 and here're some changes notes: change from V6-->V7: - rename variable - use HAS_POWER_WELL instead of IS_HASWELL - put structure inside drm_i915_private - use WARN_ON for global pointer check change from V5-->V6: - Remove duplication code in new introduced probe work - move duplication code in azx_probe_continue - remove unused #ifdef - replace request_module with symbol_request - replace spin_lock_irq with spin_lock_irqsave in gfx side - other typo fixes (review by Takashi) change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Takashi Iwai (1): ALSA: hda - Move azx_first_init() into azx_probe_continue() Wang Xingchao (3): ALSA: hda - Fix runtime PM check ALSA: hda - Add power-welll support for haswell HDA i915/drm: Add private api for power well usage drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 ++++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 81 ++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++++ sound/pci/hda/Kconfig | 10 +++++ sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 87 ++++++++++++++++++++++++++++++-------- 10 files changed, 324 insertions(+), 24 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

7 40

[alsa-devel] unsubscribe my ID
by Reena Chauhan Tomar 25 Jul '13

25 Jul '13

Kindly unsubscribe this ID <reena.nith(a)gmail.com> from your mailing list. -- Reena Tomar Scientist-C Application & Baseband Platform, SBSG ,DEAL,Raipur Road Dehradun-248001

1 0

[alsa-devel] usb-audio: capture/playback break (implicit feedback related?)
by Eldad Zack 24 Jul '13

24 Jul '13

Hi, I'm still trying to figure out the issue with the capture and playback breaking with implicit feedback. I've disabled setting the sync ep for my device and did the following: start capture start playback stop playback Now, after a while, capture stops. I sniffed the usb traffic, and saw this: When I stop the traffic, the driver sets the alternate setting to 0 on the playback interface (2 for my device). So far so good. The first iso inbound packet has data in it, but starting from the second packet, it's all zero (in length). After a while, the device stops sending packets completely. The driver still tries, and evetually sets the alternate setting to 0 on the capture interface (3 for my device). Again, there is no implicit feedback involved here. I also ruled out playback interface being put into altsetting 1 (without us knowing it) when the capture was put into altsetting 1 by doing a GET_INTERFACE in set_format(). I am not sure if this an M-Audio issue, just my specific device (Fast Track C400) or maybe it appears in other devices. Anyone has an idea why this happens? I'd appreciate if anyone can test this on other devices. Clemens, since you recently added support for Roland devices that use implicit feedback, would you please test it and see if you get the same results, if you have time? If it helps, I can send the capture. Cheers, Eldad P.S. GET_INTERFACE adapted from drivers/usb/misc/usbtest.c: static int snd_usb_get_interface(struct usb_device *dev, struct usb_interface *iface) { int ret; char buf; ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), USB_REQ_GET_INTERFACE, USB_DIR_IN|USB_RECIP_INTERFACE, 0, iface->altsetting[0].desc.bInterfaceNumber, &buf, sizeof(buf), USB_CTRL_GET_TIMEOUT); if (ret == sizeof(buf)) return buf; return -ERANGE; }

1 0

[alsa-devel] [PATCH] fix error returned by compress_wait()
by Vinod Koul 24 Jul '13

24 Jul '13

From: Eric Laurent <elaurent(a)google.com> oops() expects positive error codes. Signed-off-by: Eric Laurent <elaurent(a)google.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index 734fbbf..0e71c28 100644 --- a/compress.c +++ b/compress.c @@ -623,7 +623,7 @@ int compress_wait(struct compress *compress, int timeout_ms) return 0; } if (fds.revents & POLLERR) { - return oops(compress, -EIO, "poll returned error!"); + return oops(compress, EIO, "poll returned error!"); } return ret; } -- 1.7.0.4

1 0

[alsa-devel] [PATCH 00/27] devm cleanup, part #1, take #3
by Wolfram Sang 24 Jul '13

24 Jul '13

Here is another bit of cleaning up the devm usage. It is again removing the resource check with devm_ioremap_resource, because a) new drivers came in and b) coccinelle had a bug and missed to find a couple of occasions. Unlike last time, I think it is better if these patches go in via the subsystem trees to reduce merge conflicts. And there is one driver which I fixed manually because the original code needed some bigger update. All is based on v3.11-rc2 and the branch can be found at: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git devm_no_resource_check Other things which happened: I wanted to get rid of devm_request_and_ioremap, luckily other people are already working on it. Hooray! I already sent a patch series picking another low hanging fruit, i.e. drivers can skip devm_pinctrl handling if they are using only the default pin setup. And not devm related, there is still my proposal to rename INIT_COMPLETION to reinit_completion, that probably needs some more persistence... Yay, so much to clean up \o/ Regards, Wolfram Wolfram Sang (27): arch/mips/lantiq/xway: don't check resource with devm_ioremap_resource drivers/amba: don't check resource with devm_ioremap_resource drivers/cpuidle: don't check resource with devm_ioremap_resource drivers/dma: don't check resource with devm_ioremap_resource drivers/gpu/host1x/drm: don't check resource with devm_ioremap_resource drivers/i2c/busses: don't check resource with devm_ioremap_resource drivers/input/serio: don't check resource with devm_ioremap_resource drivers/iommu: don't check resource with devm_ioremap_resource drivers/media/platform: don't check resource with devm_ioremap_resource drivers/memory: don't check resource with devm_ioremap_resource drivers/mtd/nand: don't check resource with devm_ioremap_resource drivers/net/ethernet/stmicro/stmmac: don't check resource with devm_ioremap_resource drivers/pci/host: don't check resource with devm_ioremap_resource drivers/pinctrl: don't check resource with devm_ioremap_resource drivers/pwm: don't check resource with devm_ioremap_resource drivers/scsi/ufs: don't check resource with devm_ioremap_resource drivers/spi: don't check resource with devm_ioremap_resource drivers/staging/imx-drm: don't check resource with devm_ioremap_resource drivers/usb/phy: don't check resource with devm_ioremap_resource drivers/watchdog: don't check resource with devm_ioremap_resource sound/soc/au1x: don't check resource with devm_ioremap_resource sound/soc/cirrus: don't check resource with devm_ioremap_resource sound/soc/nuc900: don't check resource with devm_ioremap_resource sound/soc/pxa: don't check resource with devm_ioremap_resource sound/soc/tegra: don't check resource with devm_ioremap_resource sound/soc/txx9: don't check resource with devm_ioremap_resource thermal: ti-bandgap: cleanup resource allocation arch/mips/lantiq/xway/dma.c | 4 ---- drivers/amba/tegra-ahb.c | 2 -- drivers/cpuidle/cpuidle-kirkwood.c | 3 --- drivers/dma/mmp_pdma.c | 3 --- drivers/dma/mmp_tdma.c | 3 --- drivers/gpu/host1x/drm/hdmi.c | 3 --- drivers/i2c/busses/i2c-stu300.c | 3 --- drivers/input/serio/olpc_apsp.c | 3 --- drivers/iommu/tegra-smmu.c | 2 -- drivers/media/platform/coda.c | 5 ----- drivers/memory/tegra20-mc.c | 2 -- drivers/memory/tegra30-mc.c | 2 -- drivers/mtd/nand/mxc_nand.c | 5 ----- .../net/ethernet/stmicro/stmmac/stmmac_platform.c | 3 --- drivers/pci/host/pcie-designware.c | 12 ------------ drivers/pinctrl/pinctrl-imx.c | 3 --- drivers/pinctrl/pinctrl-rockchip.c | 5 ----- drivers/pinctrl/pinctrl-u300.c | 3 --- drivers/pwm/pwm-lpc32xx.c | 3 --- drivers/pwm/pwm-renesas-tpu.c | 5 ----- drivers/scsi/ufs/ufshcd-pltfrm.c | 6 ------ drivers/spi/spi-bcm2835.c | 6 ------ drivers/staging/imx-drm/imx-tve.c | 5 ----- drivers/thermal/ti-soc-thermal/ti-bandgap.c | 20 ++++---------------- drivers/usb/phy/phy-rcar-usb.c | 5 ----- drivers/watchdog/nuc900_wdt.c | 5 ----- drivers/watchdog/ts72xx_wdt.c | 10 ---------- sound/soc/au1x/psc-ac97.c | 3 --- sound/soc/cirrus/ep93xx-ac97.c | 3 --- sound/soc/cirrus/ep93xx-i2s.c | 3 --- sound/soc/nuc900/nuc900-ac97.c | 3 --- sound/soc/pxa/mmp-sspa.c | 3 --- sound/soc/tegra/tegra20_ac97.c | 7 ------- sound/soc/txx9/txx9aclc-ac97.c | 3 --- 34 files changed, 4 insertions(+), 152 deletions(-) -- 1.7.10.4

2 12

[alsa-devel] [PATCH 1/5] ASoC: dapm: Fix return value of snd_soc_dapm_put_{volsw, enum_virt}()
by Lars-Peter Clausen 24 Jul '13

24 Jul '13

The ALSA core expect the put callback of a control to return 1 if the value of the control changed and 0 if it did not. Both snd_soc_dapm_put_volsw() and snd_soc_dapm_put_enum_virt() currently always returns 0. For both functions we already have a 'change' variable which either contains 1 or 0 depending on whether the value has changed or not, so just return that. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index deb1b3d..4097a55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2703,7 +2703,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return 0; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2831,7 +2831,6 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; - int ret = 0; int wi; if (ucontrol->value.enumerated.item[0] >= e->max) @@ -2851,7 +2850,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); -- 1.8.0

2 5

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix EAPD GPIO control for Sigmatel codecs
by Takashi Iwai 24 Jul '13

24 Jul '13

The EAPD GPIO is dynamically turned on/off for some machines with Sigmatel codecs, but this didn't work as expected, and it resulted in spontaneous lost of speaker outputs per HP plugging or power-saving. This patch fixes the bug by simply including spec->eapd_mask into spec->gpio_mask and spec->gpio_data bits. Reported-and-tested-by: Eric Shattow <lucent(a)gmail.com> Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f8359..766e567 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -417,9 +417,11 @@ static void stac_update_outputs(struct hda_codec *codec) val &= ~spec->eapd_mask; else val |= spec->eapd_mask; - if (spec->gpio_data != val) + if (spec->gpio_data != val) { + spec->gpio_data = val; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, val); + } } } @@ -3612,20 +3614,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio; int i; /* override some hints */ stac_store_hints(codec); /* set up GPIO */ - gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. * otherwise, unsol event will turn it on/off dynamically */ if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); + spec->gpio_data |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); snd_hda_gen_init(codec); @@ -3915,6 +3915,7 @@ static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + spec->gpio_mask |= spec->eapd_mask; if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; -- 1.8.3.1

2 5

[alsa-devel] [PATCHv2] ALSA: usb-audio: hiface: return correct XRUN indication
by Eldad Zack 24 Jul '13

24 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of hiface, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Cc: Antonio Ospite <ospite(a)studenti.unina.it> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- * Corrected commit message sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2..c21a3df 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; -- 1.8.1.5

2 1

[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 24 Jul '13

24 Jul '13

The following changes since commit 3b2f64d00c46e1e4e9bd0bb9bb12619adac27a4b: Linux 3.11-rc2 (2013-07-21 12:05:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc2 for you to fetch changes up to c5ae4dca11523dff15cfb1763b64a71b805976aa: Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus (2013-07-23 19:17:11 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 A few small updates again, the sgtl5000 one fixes some newly triggered issues due to some probe ordering changes which were introduced in the last merge window. ---------------------------------------------------------------- Boris BREZILLON (1): ASoC: atmel-ssc: remove clk_disable_unprepare call from critical section Chih-Chung Chang (1): ASoC: max98088 - fix element type of the register cache. Dan Carpenter (1): ASoC: core: double free in snd_soc_add_platform() H Hartley Sweeten (1): ASoC: ep93xx: fix build of ep93xx-ac97.c Mark Brown (6): Merge remote-tracking branch 'asoc/fix/atmel' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/ep93xx' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98088' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus Richard Zhao (1): ASoC: tegra: correct playback_dma_data setup Shawn Guo (1): ASoC: sgtl5000: defer the probe if clock is not found drivers/misc/atmel-ssc.c | 11 ++++++++--- sound/soc/cirrus/ep93xx-ac97.c | 4 ++-- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/sgtl5000.c | 3 +++ sound/soc/soc-core.c | 4 +--- sound/soc/tegra/tegra20_ac97.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 4 ++-- 7 files changed, 20 insertions(+), 14 deletions(-)

2 1

[alsa-devel] [PATCH 1/4] ASOC: tegra: move AC97 clock defines to the controller node
by Lucas Stach 24 Jul '13

24 Jul '13

Different from other Tegra sound controllers drivers, the AC97 controller driver uses the tegra asoc utils directly to request the needed clocks, as they are needed at AC97 init time. Move the DT clock defines to the right place. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> --- CC: <alsa-devel(a)alsa-project.org> CC: <broonie(a)kernel.org> --- arch/arm/boot/dts/tegra20-colibri-512.dtsi | 5 ----- arch/arm/boot/dts/tegra20.dtsi | 6 +++++- sound/soc/tegra/tegra20_ac97.c | 2 +- 3 files changed, 6 insertions(+), 7 deletions(-) diff --git a/arch/arm/boot/dts/tegra20-colibri-512.dtsi b/arch/arm/boot/dts/tegra20-colibri-512.dtsi index 2fcb3f2..fbb52e0 100644 --- a/arch/arm/boot/dts/tegra20-colibri-512.dtsi +++ b/arch/arm/boot/dts/tegra20-colibri-512.dtsi @@ -491,11 +491,6 @@ "Mic", "MIC1"; nvidia,ac97-controller = <&ac97>; - - clocks = <&tegra_car TEGRA20_CLK_PLL_A>, - <&tegra_car TEGRA20_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA20_CLK_CDEV1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; }; regulators { diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi index 9653fd8..ad13f57 100644 --- a/arch/arm/boot/dts/tegra20.dtsi +++ b/arch/arm/boot/dts/tegra20.dtsi @@ -223,7 +223,11 @@ reg = <0x70002000 0x200>; interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; nvidia,dma-request-selector = <&apbdma 12>; - clocks = <&tegra_car TEGRA20_CLK_AC97>; + clocks = <&tegra_car TEGRA20_CLK_PLL_A>, + <&tegra_car TEGRA20_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA20_CLK_CDEV1>, + <&tegra_car TEGRA20_CLK_AC97>; + clock-names = "pll_a", "pll_a_out0", "mclk", "ac97"; status = "disabled"; }; diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 6c48662..6bbffd1 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -326,7 +326,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ac97); - ac97->clk_ac97 = devm_clk_get(&pdev->dev, NULL); + ac97->clk_ac97 = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(ac97->clk_ac97)) { dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); ret = PTR_ERR(ac97->clk_ac97); -- 1.8.3.1

3 7

Re: [alsa-devel] [PATCH] ALSA: SoC: Added jack detection kcontrol support
by Mark Brown 24 Jul '13

24 Jul '13

On Fri, Jul 19, 2013 at 05:59:22PM -0700, Felipe F. Tonello wrote: > From: "Felipe F. Tonello" <ftonello(a)cercacor.com> > > It adds soc jack support for alsa kcontrol. > > This support is necessary since the new kcontrol is used by user-space > daemons, such as PulseAudio(>=2.0), to do jack detection. > > Signed-off-by: Felipe F. Tonello <ftonello(a)cercacor.com> > --- > include/sound/soc.h | 1 + > sound/soc/Kconfig | 1 + > sound/soc/soc-jack.c | 22 ++++++++++++++++++++++ > 3 files changed, 24 insertions(+) No, this makes no sense at all as an ASoC specific feature - it should be a standard feature of ALSA jacks. The whole point of having ALSA core infrastructure for jacks is to avoid having to go round doing make work stuff like this. The fact that the kcontrols were introduced as a HDA-specific feature at all is of course a problem here...

3 5

[alsa-devel] question about alsa-lib commit 8c9e4114 (Add struct timeval and timespec definition when _POSIX_C_SOURCE is not defined)
by John Spencer 24 Jul '13

24 Jul '13

the commit http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=8c9e4114a23aad96… was added in 2006 with a FIXME hint that it could cause a build error. this is exactly what happens when using the header with musl libc to build software depending on alsa-lib (for example "sweep"). may i ask what for this hack was added ? i suspect it's for supporting some ancient version of glibc that didnt define those structs. in that case something like #if defined(__GLIBC__) && (__GLIBC__ < 2) would probably make more sense. i could even imagine that the target for the hack is no longer supported anyway, so i think it could simply be removed. the problematic part is that it's the job of the *application* to request the POSIX namespace, not the job of the libc. since sweep's build environment does not set _POSIX_SOURCE, the code gets activated and causes a conflict. --JS p.s. thanks for applying the signal.h patch!

1 0

[alsa-devel] [PATCH] ASoC: Remove unused dapm_get_snd_card() and dapm_get_soc_card()
by Lars-Peter Clausen 24 Jul '13

24 Jul '13

These two functions were added two years ago in commit 4805608 ("ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.") but have remained unused so far. Considering that the dapm context actually has a direct pointer to the card the functions also seem to be unnecessary. E.g. the expressions 'dapm_get_soc_card(dapm)' and 'dapm->card' yield the same result. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 30 ------------------------------ 1 file changed, 30 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 346ac91..deb1b3d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -174,36 +174,6 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } -/* get snd_card from DAPM context */ -static inline struct snd_card *dapm_get_snd_card( - struct snd_soc_dapm_context *dapm) -{ - if (dapm->codec) - return dapm->codec->card->snd_card; - else if (dapm->platform) - return dapm->platform->card->snd_card; - else - BUG(); - - /* unreachable */ - return NULL; -} - -/* get soc_card from DAPM context */ -static inline struct snd_soc_card *dapm_get_soc_card( - struct snd_soc_dapm_context *dapm) -{ - if (dapm->codec) - return dapm->codec->card; - else if (dapm->platform) - return dapm->platform->card; - else - BUG(); - - /* unreachable */ - return NULL; -} - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; -- 1.8.0

4 8

[alsa-devel] [PATCH] ASoC: Samsung: Modify the I2S driver to support I2S on Exynos5420
by Padmavathi Venna 23 Jul '13

23 Jul '13

Exynos5420 added support for I2S TDM mode. For this, there are some register changes in the I2S controller. This patch adds the relevant register changes to support I2S in normal mode. This patch adds a quirk for TDM mode and if TDM mode is present all the relevent changes will be applied. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> [abrestic: style cleanup and documentation] Signed-off-by: Andrew Bresticker <abrestic(a)chromium.org> Reviewed-on: https://gerrit-int.chromium.org/37840 Reviewed-by: Simon Glass <sjg(a)google.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 2 + include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 51 ++++++--- sound/soc/samsung/i2s.c | 117 ++++++++++++++++---- 4 files changed, 132 insertions(+), 39 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 025e66b..b8593d5 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -28,6 +28,8 @@ Optional SoC Specific Properties: enabled or disabled based on need. - samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, then this flag is enabled. +- samsung,supports-tdm: If the I2S controller supports TDM, then this flag + must be enabled. - samsung,idma-addr: Internal DMA register base address of the audio sub system(used in secondary sound source). - pinctrl-0: Should specify pin control groups used for this controller. diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 8827259..9efc04d 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -36,6 +36,7 @@ struct samsung_i2s { */ #define QUIRK_NO_MUXPSR (1 << 2) #define QUIRK_NEED_RSTCLR (1 << 3) +#define QUIRK_SUPPORTS_TDM (1 << 4) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index c0e6d9a..821a502 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -31,6 +31,10 @@ #define I2SLVL1ADDR 0x34 #define I2SLVL2ADDR 0x38 #define I2SLVL3ADDR 0x3c +#define I2SSTR1 0x40 +#define I2SVER 0x44 +#define I2SFIC2 0x48 +#define I2STDM 0x4c #define CON_RSTCLR (1 << 31) #define CON_FRXOFSTATUS (1 << 26) @@ -95,24 +99,39 @@ #define MOD_RXONLY (1 << 8) #define MOD_TXRX (2 << 8) #define MOD_MASK (3 << 8) -#define MOD_LR_LLOW (0 << 7) -#define MOD_LR_RLOW (1 << 7) -#define MOD_SDF_IIS (0 << 5) -#define MOD_SDF_MSB (1 << 5) -#define MOD_SDF_LSB (2 << 5) -#define MOD_SDF_MASK (3 << 5) -#define MOD_RCLK_256FS (0 << 3) -#define MOD_RCLK_512FS (1 << 3) -#define MOD_RCLK_384FS (2 << 3) -#define MOD_RCLK_768FS (3 << 3) -#define MOD_RCLK_MASK (3 << 3) -#define MOD_BCLK_32FS (0 << 1) -#define MOD_BCLK_48FS (1 << 1) -#define MOD_BCLK_16FS (2 << 1) -#define MOD_BCLK_24FS (3 << 1) -#define MOD_BCLK_MASK (3 << 1) +#define MOD_LRP_SHIFT 7 +#define MOD_LR_LLOW 0 +#define MOD_LR_RLOW 1 +#define MOD_SDF_SHIFT 5 +#define MOD_SDF_IIS 0 +#define MOD_SDF_MSB 1 +#define MOD_SDF_LSB 2 +#define MOD_SDF_MASK 3 +#define MOD_RCLK_SHIFT 3 +#define MOD_RCLK_256FS 0 +#define MOD_RCLK_512FS 1 +#define MOD_RCLK_384FS 2 +#define MOD_RCLK_768FS 3 +#define MOD_RCLK_MASK 3 +#define MOD_BCLK_SHIFT 1 +#define MOD_BCLK_32FS 0 +#define MOD_BCLK_48FS 1 +#define MOD_BCLK_16FS 2 +#define MOD_BCLK_24FS 3 +#define MOD_BCLK_MASK 3 #define MOD_8BIT (1 << 0) +#define EXYNOS5420_MOD_LRP_SHIFT 15 +#define EXYNOS5420_MOD_SDF_SHIFT 6 +#define EXYNOS5420_MOD_RCLK_SHIFT 4 +#define EXYNOS5420_MOD_BCLK_SHIFT 0 +#define EXYNOS5420_MOD_BCLK_64FS 4 +#define EXYNOS5420_MOD_BCLK_96FS 5 +#define EXYNOS5420_MOD_BCLK_128FS 6 +#define EXYNOS5420_MOD_BCLK_192FS 7 +#define EXYNOS5420_MOD_BCLK_256FS 8 +#define EXYNOS5420_MOD_BCLK_MASK 0xf + #define MOD_CDCLKCON (1 << 12) #define PSR_PSREN (1 << 15) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 3fcf8d7..398f8db 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -198,7 +198,13 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { - u32 rfs = (readl(i2s->addr + I2SMOD) >> 3) & 0x3; + u32 rfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs = readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT; + rfs &= MOD_RCLK_MASK; switch (rfs) { case 3: return 768; @@ -212,21 +218,26 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { u32 mod = readl(i2s->addr + I2SMOD); + int rfs_shift; - mod &= ~MOD_RCLK_MASK; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs_shift = MOD_RCLK_SHIFT; + mod &= ~(MOD_RCLK_MASK << rfs_shift); switch (rfs) { case 768: - mod |= MOD_RCLK_768FS; + mod |= (MOD_RCLK_768FS << rfs_shift); break; case 512: - mod |= MOD_RCLK_512FS; + mod |= (MOD_RCLK_512FS << rfs_shift); break; case 384: - mod |= MOD_RCLK_384FS; + mod |= (MOD_RCLK_384FS << rfs_shift); break; default: - mod |= MOD_RCLK_256FS; + mod |= (MOD_RCLK_256FS << rfs_shift); break; } @@ -236,9 +247,22 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { - u32 bfs = (readl(i2s->addr + I2SMOD) >> 1) & 0x3; + u32 bfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT; + bfs &= EXYNOS5420_MOD_BCLK_MASK; + } else { + bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; + bfs &= MOD_BCLK_MASK; + } switch (bfs) { + case 8: return 256; + case 7: return 192; + case 6: return 128; + case 5: return 96; + case 4: return 64; case 3: return 24; case 2: return 16; case 1: return 48; @@ -250,21 +274,50 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { u32 mod = readl(i2s->addr + I2SMOD); + int bfs_shift; + int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; - mod &= ~MOD_BCLK_MASK; + if (tdm) { + bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT; + mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift); + } else { + bfs_shift = MOD_BCLK_SHIFT; + mod &= ~(MOD_BCLK_MASK << bfs_shift); + } + + /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ + if (!tdm && bfs > 48) { + dev_err(&i2s->pdev->dev, "Unsupported BCLK divider\n"); + return; + } switch (bfs) { case 48: - mod |= MOD_BCLK_48FS; + mod |= (MOD_BCLK_48FS << bfs_shift); break; case 32: - mod |= MOD_BCLK_32FS; + mod |= (MOD_BCLK_32FS << bfs_shift); break; case 24: - mod |= MOD_BCLK_24FS; + mod |= (MOD_BCLK_24FS << bfs_shift); break; case 16: - mod |= MOD_BCLK_16FS; + mod |= (MOD_BCLK_16FS << bfs_shift); + break; + case 64: + mod |= (EXYNOS5420_MOD_BCLK_64FS << bfs_shift); + break; + case 96: + mod |= (EXYNOS5420_MOD_BCLK_96FS << bfs_shift); + break; + case 128: + mod |= (EXYNOS5420_MOD_BCLK_128FS << bfs_shift); + break; + case 192: + mod |= (EXYNOS5420_MOD_BCLK_192FS << bfs_shift); + break; + case 256: + mod |= (EXYNOS5420_MOD_BCLK_256FS << bfs_shift); break; default: dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n"); @@ -492,19 +545,31 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, struct i2s_dai *i2s = to_info(dai); u32 mod = readl(i2s->addr + I2SMOD); u32 tmp = 0; + int lrp_shift, sdf_shift, sdf_mask, lrp_rlow; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + lrp_shift = EXYNOS5420_MOD_LRP_SHIFT; + sdf_shift = EXYNOS5420_MOD_SDF_SHIFT; + } else { + lrp_shift = MOD_LRP_SHIFT; + sdf_shift = MOD_SDF_SHIFT; + } + + sdf_mask = MOD_SDF_MASK << sdf_shift; + lrp_rlow = MOD_LR_RLOW << lrp_shift; /* Format is priority */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_MSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_MSB << sdf_shift); break; case SND_SOC_DAIFMT_LEFT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_LSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_LSB << sdf_shift); break; case SND_SOC_DAIFMT_I2S: - tmp |= MOD_SDF_IIS; + tmp |= (MOD_SDF_IIS << sdf_shift); break; default: dev_err(&i2s->pdev->dev, "Format not supported\n"); @@ -519,10 +584,10 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_NB_IF: - if (tmp & MOD_LR_RLOW) - tmp &= ~MOD_LR_RLOW; + if (tmp & lrp_rlow) + tmp &= ~lrp_rlow; else - tmp |= MOD_LR_RLOW; + tmp |= lrp_rlow; break; default: dev_err(&i2s->pdev->dev, "Polarity not supported\n"); @@ -544,15 +609,18 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } + /* + * Don't change the I2S mode if any controller is active on this + * channel. + */ if (any_active(i2s) && - ((mod & (MOD_SDF_MASK | MOD_LR_RLOW - | MOD_SLAVE)) != tmp)) { + ((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; } - mod &= ~(MOD_SDF_MASK | MOD_LR_RLOW | MOD_SLAVE); + mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE); mod |= tmp; writel(mod, i2s->addr + I2SMOD); @@ -1170,6 +1238,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (of_find_property(np, "samsung,supports-rstclr", NULL)) quirks |= QUIRK_NEED_RSTCLR; + if (of_find_property(np, "samsung,supports-tdm", NULL)) + quirks |= QUIRK_SUPPORTS_TDM; + if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { if (quirks & QUIRK_SEC_DAI) { -- 1.7.4.4

4 16

[alsa-devel] [PATCH 1/2] ASoC: Build adau1701 when SND_SOC_ALL_CODECS is selected
by Lars-Peter Clausen 23 Jul '13

23 Jul '13

The adau1701 driver was removed from SND_SOC_ALL_CODECS in commit commit ee8c7e9 ("ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency") due to the dependency on the SigmaDSP firmware loader which was only available on a limited set of platforms. This was fixed quite some time ago in commit 40216ce7 ("ASoC: Move SigmaDSP firmware loader to ASoC") though, so we can add the driver back again to SND_SOC_ALL_CODECS. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 63efc4e..d038a58 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- 1.8.0

2 2

[alsa-devel] [PATCH v3] ARM: kirkwood: extend the kirkwood i2s driver for DT usage
by Jean-Francois Moine 23 Jul '13

23 Jul '13

The kirkwood i2s driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3 - change compatible from kirkwood to mvebu (Andrew Lunn) v2 - this patch replaces the previous [PATCH 2/2] ARM: kirkwood: extend the kirkwood i2s driver for DT usage - 2 possible clocks (Sebastian Hesselbarth) - shorter io mapping (Andrew Lunn) - less #ifdef's --- Documentation/devicetree/bindings/sound/kirkwood-i2s.txt | 24 +++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 58 ++++++++++++++++++++------ 2 files changed, 70 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt new file mode 100644 index 0000000..8f1e534 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt @@ -0,0 +1,24 @@ +* mvebu (Kirkwood, Dove, Armada 370) I2S controller + +Required properties: + +- compatible: "marvell,mvebu-i2s" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-i2s"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..019e340 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -12,7 +12,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/io.h> #include <linux/slab.h> #include <linux/mbus.h> @@ -22,6 +21,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "kirkwood-i2s" @@ -461,6 +462,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -481,24 +483,45 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); - return -EINVAL; + /* get the DT or static parameters */ + np = pdev->dev.of_node; + if (np) { + struct of_phandle_args clkspec; + + priv->burst = 128; /* might be 32 or 128 */ + priv->clk = of_clk_get(np, 0); /* internal clock */ + err = of_parse_phandle_with_args(np, + "clocks", "#clock-cells", 1, + &clkspec); + if (err) { + priv->extclk = ERR_PTR(-EINVAL); /* no external clock */ + } else { + priv->extclk = of_clk_get(np, 1); + if (IS_ERR(priv->extclk)) { + err = -EPROBE_DEFER; + goto fail; + } + } + } else { + if (!data) { + dev_err(&pdev->dev, "no platform data ?!\n"); + return -EINVAL; + } + priv->burst = data->burst; + priv->clk = clk_get(&pdev->dev, NULL); + priv->extclk = clk_get(&pdev->dev, "extclk"); } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); - return PTR_ERR(priv->clk); + err = PTR_ERR(priv->clk); + goto fail; } err = clk_prepare_enable(priv->clk); if (err < 0) - return err; + goto fail; - priv->extclk = clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { clk_put(priv->extclk); @@ -515,7 +538,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -528,12 +551,13 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!err) return 0; dev_err(&pdev->dev, "snd_soc_register_component failed\n"); - +fail: if (!IS_ERR(priv->extclk)) { clk_disable_unprepare(priv->extclk); clk_put(priv->extclk); } clk_disable_unprepare(priv->clk); + clk_put(priv->clk); return err; } @@ -549,16 +573,26 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) clk_put(priv->extclk); } clk_disable_unprepare(priv->clk); + clk_put(priv->clk); return 0; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "marvell,mvebu-i2s" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

6 15

[alsa-devel] [PATCH 2/4] ASOC: tegra: fix matching of AC97 components
by Lucas Stach 23 Jul '13

23 Jul '13

Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> --- This got broken some time ago. No cc stable as it's not really a regression, as I can't remember a single point in time when everything AC97 related would have been in place and working. CC: <alsa-devel(a)alsa-project.org> CC: <broonie(a)kernel.org> --- sound/soc/tegra/tegra_wm9712.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e11963..45b5789 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- 1.8.3.1

2 1

[alsa-devel] alsa-lib 1.0.27 incorrect definition of __kernel_off_t?
by Kevin Groeneveld 23 Jul '13

23 Jul '13

I tried doing a cross compile of alsa-lib 1.0.27.2 today and ended up with the following error: error: conflicting types for 'off_t' Commit 9b5beef1 adds the following line to local.h: #define __kernel_off_t off_t Not only is __kernel_off_t already defined elsewhere in my header files, the above line tries to define __kernel_off_t (which I think should be 32 bit on my platform) to off_t which is 64 bit. I just deleted the line and everything seems to be working so far. Kevin

2 2

[alsa-devel] [PATCH] ARM: kirkwood: enable S/PDIF
by Jean-Francois Moine 23 Jul '13

23 Jul '13

This patch enables S/PDIF. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..ad94c50 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -159,6 +159,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | + KIRKWOOD_PLAYCTL_SPDIF_EN | KIRKWOOD_PLAYCTL_I2S_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | KIRKWOOD_RECCTL_I2S_EN; @@ -169,6 +170,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_20; ctl_play = KIRKWOOD_PLAYCTL_SIZE_20 | + KIRKWOOD_PLAYCTL_SPDIF_EN | KIRKWOOD_PLAYCTL_I2S_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_20 | KIRKWOOD_RECCTL_I2S_EN; @@ -177,6 +179,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | + KIRKWOOD_PLAYCTL_SPDIF_EN | KIRKWOOD_PLAYCTL_I2S_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | KIRKWOOD_RECCTL_I2S_EN; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

4 7

[alsa-devel] [PATCH] ARM: kirkwood: extend the kirkwood pcm/dma driver for DT usage
by Jean-Francois Moine 23 Jul '13

23 Jul '13

The kirkwood pcm/dma driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Documentation/devicetree/bindings/sound/kirkwood-dma.txt | 11 +++++++++++ sound/soc/kirkwood/kirkwood-dma.c | 11 +++++++++++ 2 files changed, 22 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/kirkwood-dma.txt b/Documentation/devicetree/bindings/sound/kirkwood-dma.txt new file mode 100644 index 0000000..9dd2989 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-dma.txt @@ -0,0 +1,11 @@ +* mvebu (Kirkwood, Dove, Armada 370) PCM/DMA controller + +Required properties: + +- compatible: "marvell,mvebu-pcm-audio" + +Example: + +pcm: pcm { + compatible = "marvell,mvebu-pcm-audio"; +}; diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index a9f1453..b91045a 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -19,6 +19,8 @@ #include <linux/dma-mapping.h> #include <linux/mbus.h> #include <sound/soc.h> +#include <linux/of.h> + #include "kirkwood.h" #define KIRKWOOD_RATES \ @@ -383,10 +385,19 @@ static int kirkwood_soc_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_pcm_of_match[] = { + { .compatible = "marvell,mvebu-pcm-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_pcm_of_match); +#endif + static struct platform_driver kirkwood_pcm_driver = { .driver = { .name = "kirkwood-pcm-audio", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_pcm_of_match), }, .probe = kirkwood_soc_platform_probe, -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 3

[alsa-devel] [PATCH] alsa-utils: fix wrong alias for signal.h
by John Spencer 23 Jul '13

23 Jul '13

in POSIX, there is no <sys/signal.h>. it's merely a legacy alias used by glibc, and the header does nothing else than including <signal.h>. so let's do the right thing and use the right name, which works everywhere. Signed-Off-By: John Spencer <maillist-alsa(a)barfooze.de>

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: hiface: return correct XRUN indication
by Eldad Zack 23 Jul '13

23 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2..c21a3df 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; -- 1.8.1.5

3 3

[alsa-devel] [PATCH] ASoC: samsung: Enable COMPILE_TEST builds
by Mark Brown 22 Jul '13

22 Jul '13

From: Mark Brown <broonie(a)linaro.org> Platform using OF and some others don't have any dependencies on arch/arm so can be built on any platform. Enable that for better compile coverage when COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/Kconfig | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9855dfc..df28dc8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on PLAT_SAMSUNG + depends on PLAT_SAMSUNG || COMPILE_TEST select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help @@ -150,7 +150,7 @@ config SND_SOC_SMARTQ config SND_SOC_GONI_AQUILA_WM8994 tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" - depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) + depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA || COMPILE_TEST) depends on I2C=y && GENERIC_HARDIRQS select SND_SAMSUNG_I2S select MFD_WM8994 @@ -186,7 +186,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -195,13 +195,13 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM8962 config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -211,14 +211,14 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && (MACH_WLF_CRAGG_6410 || COMPILE_TEST) select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: mxs: Depends on COMMON_CLK
by Mark Brown 22 Jul '13

22 Jul '13

From: Mark Brown <broonie(a)linaro.org> The SAIF driver is a clock provider so specifically needs the common clock implementedation. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 7daf860..219235c 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

2 1

Re: [alsa-devel] Samsung S5PV210 I2S DMA
by Mike Dyer 22 Jul '13

22 Jul '13

On Sat, 2013-07-06 at 16:22 +0600, Alexey Galakhov wrote: > Hi, > > > The DMA engine emits a lot of messages complaining of (I think) a read > > fault: > > > dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 > > dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 > > I'm debugging the same error right now. FTC-40000 means "Read error", > i.e. invalid address of the DMA source block. > > Regards, > Alex In my case it turned out the TZPC hadn't been configured properly by the bootloader. I had to modify to the bootloader to make sure the I2S was non secured (it's secure by default). Hope this helps, Mike

2 1

[alsa-devel] [patch] ASoC: core: double free in snd_soc_add_platform()
by Dan Carpenter 22 Jul '13

22 Jul '13

There are three callers for this function, and none of them want it to free platform for them. It leads to a double free. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070c..d82ee38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3908,10 +3908,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) { - kfree(platform); + if (platform->name == NULL) return -ENOMEM; - } platform->dev = dev; platform->driver = platform_drv;

2 1

[alsa-devel] [PATCH] ASoC: mxs: Remove unneeded mach-types.h inclusions
by Mark Brown 22 Jul '13

22 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54511c5..b56b8a0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -31,7 +31,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> #include "mxs-saif.h" diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 1b134d7..b2e372d 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -25,7 +25,6 @@ #include <sound/soc.h> #include <sound/jack.h> #include <sound/soc-dapm.h> -#include <asm/mach-types.h> #include "../codecs/sgtl5000.h" #include "mxs-saif.h" -- 1.8.3.2

2 1

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Daniel Mack 22 Jul '13

22 Jul '13

Hi, On 20.07.2013 19:25, Jean-Francois Moine wrote: > This generic simple card driver uses DT values to instanciate an audio > system in which the real work is done by the subdrivers (audio > controller, audio codec and pcm/dma controller). > > Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> > --- > diff --git a/sound/soc/generic/simple-dt-card.c b/sound/soc/generic/simple-dt-card.c > new file mode 100644 There is a simple-card driver in the tree already, and there were several attempts to add DT bindings for it in the past - have you seen that? Search for "ASoC: add simple-card DT support" in the archives ... Daniel

3 3

[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Check the return value from clk_prepare_enable()
by Fabio Estevam 22 Jul '13

22 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e260f1f..1a5da1e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -73,8 +73,11 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); @@ -224,14 +227,19 @@ EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, unsigned int pdcr) { + int ret; + if (audmux_type != IMX31_AUDMUX) return -EINVAL; if (!audmux_base) return -ENOSYS; - if (audmux_clk) - clk_prepare_enable(audmux_clk); + if (audmux_clk) { + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; + } writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Support SND_SOC_DAIFMT_CBM_CFS for omap3/4
by Michael Trimarchi 22 Jul '13

22 Jul '13

Add SND_SOC_DAIFMT_CBM_CFS support for omap3/omap4. The patch was tested on a pandaboard-es board connected to the pcm1792a codec. mcbspx_fsx must configured as output and mcbspx_clkx must be configured as input. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..6c19bba 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -433,6 +433,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* Sample rate generator drives the FS */ regs->srgr2 |= FSGM; break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP slave. FS clock as output */ + regs->srgr2 |= FSGM; + regs->pcr0 |= FSXM; + break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ break; -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Mark Brown 22 Jul '13

22 Jul '13

On Sun, Jul 21, 2013 at 10:34:09AM +0800, Richard Zhao wrote: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct Applied, thanks.

1 0

[alsa-devel] [PATCH 1/3] ASoC: fsl: fsl_ssi: Add MODULE_ALIAS
by Fabio Estevam 22 Jul '13

22 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add MODULE_ALIAS, so that auto module loading can work. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c9974a4..429fbfd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,6 +900,7 @@ static struct platform_driver fsl_ssi_driver = { module_platform_driver(fsl_ssi_driver); +MODULE_ALIAS("platform:fsl-ssi-dai"); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL v2"); -- 1.8.1.2

3 6

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Mark Brown 22 Jul '13

22 Jul '13

On Sat, Jul 20, 2013 at 07:25:54PM +0200, Jean-Francois Moine wrote: > Documentation/devicetree/bindings/sound/simple-dt-card.txt | 18 ++++ > sound/soc/generic/Kconfig | 7 ++ > sound/soc/generic/Makefile | 2 + > sound/soc/generic/simple-dt-card.c | 137 ++++++++++++++++++++++++ > 4 files changed, 163 insertions(+) The obvious question here is why is this a separate driver to the existing simple card driver rather than DT bindings for that? I don't see that mentioned here.

1 0

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Lucas Stach 21 Jul '13

21 Jul '13

Am Sonntag, den 21.07.2013, 10:34 +0800 schrieb Richard Zhao: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct > > It also corrects slave_id initialization in tegra20_ac97 driver. > > Signed-off-by: Richard Zhao <rizhao(a)nvidia.com> > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: <stable(a)vger.kernel.org> # 3.10 Acked-by: Lucas Stach <dev(a)lynxeye.de> > --- > sound/soc/tegra/tegra20_ac97.c | 6 +++--- > sound/soc/tegra/tegra20_spdif.c | 4 ++-- > 2 files changed, 5 insertions(+), 5 deletions(-) > > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > index e58233f..6c48662 100644 > --- a/sound/soc/tegra/tegra20_ac97.c > +++ b/sound/soc/tegra/tegra20_ac97.c > @@ -389,9 +389,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) > ac97->capture_dma_data.slave_id = of_dma[1]; > > ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; > - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > - ac97->capture_dma_data.maxburst = 4; > - ac97->capture_dma_data.slave_id = of_dma[0]; > + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > + ac97->playback_dma_data.maxburst = 4; > + ac97->playback_dma_data.slave_id = of_dma[1]; > > ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); > if (ret) > diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c > index 5eaa12c..551b3c9 100644 > --- a/sound/soc/tegra/tegra20_spdif.c > +++ b/sound/soc/tegra/tegra20_spdif.c > @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) > } > > spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; > - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > - spdif->capture_dma_data.maxburst = 4; > + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > + spdif->playback_dma_data.maxburst = 4; > spdif->playback_dma_data.slave_id = dmareq->start; > > pm_runtime_enable(&pdev->dev);

1 0

Re: [alsa-devel] [PATCH] sound: replace strict_strto*() with kstrto*()
by Takashi Iwai 21 Jul '13

21 Jul '13

At Fri, 19 Jul 2013 16:24:59 +0900, Jingoo Han wrote: > > The usage of strict_strto*() is not preferred, because > strict_strto*() is obsolete. Thus, kstrto*() should be > used. > > Signed-off-by: Jingoo Han <jg1.han(a)samsung.com> Thanks, applied. Takashi > --- > sound/drivers/dummy.c | 2 +- > sound/pci/hda/hda_hwdep.c | 6 +++--- > sound/soc/codecs/wm8962.c | 2 +- > sound/soc/omap/mcbsp.c | 2 +- > sound/soc/soc-core.c | 8 +++++--- > 5 files changed, 11 insertions(+), 9 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index 11048cc..915b4d7 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry, > if (i >= ARRAY_SIZE(fields)) > continue; > snd_info_get_str(item, ptr, sizeof(item)); > - if (strict_strtoull(item, 0, &val)) > + if (kstrtoull(item, 0, &val)) > continue; > if (fields[i].size == sizeof(int)) > *get_dummy_int_ptr(dummy, fields[i].offset) = val; > diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c > index ce67608..fe0bda1 100644 > --- a/sound/pci/hda/hda_hwdep.c > +++ b/sound/pci/hda/hda_hwdep.c > @@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev, \ > struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ > struct hda_codec *codec = hwdep->private_data; \ > unsigned long val; \ > - int err = strict_strtoul(buf, 0, &val); \ > + int err = kstrtoul(buf, 0, &val); \ > if (err < 0) \ > return err; \ > codec->type = val; \ > @@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) > p = snd_hda_get_hint(codec, key); > if (!p) > ret = -ENOENT; > - else if (strict_strtoul(p, 0, &val)) > + else if (kstrtoul(p, 0, &val)) > ret = -EINVAL; > else { > *valp = val; > @@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ > struct hda_codec **codecp) \ > { \ > unsigned long val; \ > - if (!strict_strtoul(buf, 0, &val)) \ > + if (!kstrtoul(buf, 0, &val)) \ > (*codecp)->name = val; \ > } > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > index e2de9ec..e37c06f 100644 > --- a/sound/soc/codecs/wm8962.c > +++ b/sound/soc/codecs/wm8962.c > @@ -3175,7 +3175,7 @@ static ssize_t wm8962_beep_set(struct device *dev, > long int time; > int ret; > > - ret = strict_strtol(buf, 10, &time); > + ret = kstrtol(buf, 10, &time); > if (ret != 0) > return ret; > > diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c > index 361e4c0..83433fd 100644 > --- a/sound/soc/omap/mcbsp.c > +++ b/sound/soc/omap/mcbsp.c > @@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \ > unsigned long val; \ > int status; \ > \ > - status = strict_strtoul(buf, 0, &val); \ > + status = kstrtoul(buf, 0, &val); \ > if (status) \ > return status; \ > \ > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index 0ec070c..88daa64 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -192,7 +192,7 @@ static ssize_t pmdown_time_set(struct device *dev, > struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); > int ret; > > - ret = strict_strtol(buf, 10, &rtd->pmdown_time); > + ret = kstrtol(buf, 10, &rtd->pmdown_time); > if (ret) > return ret; > > @@ -237,6 +237,7 @@ static ssize_t codec_reg_write_file(struct file *file, > char *start = buf; > unsigned long reg, value; > struct snd_soc_codec *codec = file->private_data; > + int ret; > > buf_size = min(count, (sizeof(buf)-1)); > if (copy_from_user(buf, user_buf, buf_size)) > @@ -248,8 +249,9 @@ static ssize_t codec_reg_write_file(struct file *file, > reg = simple_strtoul(start, &start, 16); > while (*start == ' ') > start++; > - if (strict_strtoul(start, 16, &value)) > - return -EINVAL; > + ret = kstrtoul(start, 16, &value); > + if (ret) > + return ret; > > /* Userspace has been fiddling around behind the kernel's back */ > add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE); > -- > 1.7.10.4 > >

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: 6fire: return correct XRUN indication
by Eldad Zack 21 Jul '13

21 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index c5b9cac..60ac8fb 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add snd_hda_jack_detect_state() helper function
by Takashi Iwai 21 Jul '13

21 Jul '13

snd_hda_jack_detect() function returns a boolean value for a jack plugged in or not, but it also returns always true when the corresponding pin is phantom (i.e. fixed). This is OK in most cases, but it makes the generic parser misbehaving about the auto-mute or auto-mic switching, e.g. when one of headphone pins is a fixed. Namely, the driver decides whether to mute the speaker or not, just depending on the headphone plug state: if one of the headphone jacks is seen as active, then the speaker is muted. Thus this will result always in the muted speaker output. So, the problem is the function returns a boolean, after all, although we need to think of "phantom" jack. Now a new function, snd_hda_jack_detect_state() is introduced to return these tristates. The generic parser uses this function for checking the headphone or mic jack states. Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for keeping compatibility in other driver codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 +++++--- sound/pci/hda/hda_jack.c | 18 ++++++++++++------ sound/pci/hda/hda_jack.h | 13 ++++++++++++- 3 files changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..f5c2d1f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3724,7 +3724,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, /* check each pin in the given array; returns true if any of them is plugged */ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { - int i, present = 0; + int i; + bool present = false; for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; @@ -3733,7 +3734,8 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) /* don't detect pins retasked as inputs */ if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN) continue; - present |= snd_hda_jack_detect(codec, nid); + if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT) + present = true; } return present; } @@ -3887,7 +3889,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; - if (snd_hda_jack_detect(codec, pin)) { + if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; } diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..dc93761 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_pin_sense); /** - * snd_hda_jack_detect - query pin Presence Detect status + * snd_hda_jack_detect_state - query pin Presence Detect status * @codec: the CODEC to sense * @nid: the pin NID to sense * - * Query and return the pin's Presence Detect status. + * Query and return the pin's Presence Detect status, as either + * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. */ -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return get_jack_plug_state(sense); + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack && jack->phantom_jack) + return HDA_JACK_PHANTOM; + else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + return HDA_JACK_PRESENT; + else + return HDA_JACK_NOT_PRESENT; } -EXPORT_SYMBOL_HDA(snd_hda_jack_detect); +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index ec12abd..379420c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); + +/* the jack state returned from snd_hda_jack_detect_state() */ +enum { + HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM, +}; + +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); + +static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; +} bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); -- 1.8.3.1

2 2

Re: [alsa-devel] [PATCH] ASoC: tegra: correct playback_dma_data setup
by Stephen Warren 21 Jul '13

21 Jul '13

On 07/20/2013 08:34 PM, Richard Zhao wrote: > The errors were caused by copy/paste mistake in below commit > since v3.10: > 3489d50 ASoC: tegra: Use common DAI DMA data struct > > It also corrects slave_id initialization in tegra20_ac97 driver. Acked-by: Stephen Warren <swarren(a)nvidia.com> Hmm, that's what I get for only testing I2S and not SPDIF or AC'97 (not that I can test the latter).

1 0

Re: [alsa-devel] [PATCH] ASoC: generic: add simple card with devicetree support
by Stephen Warren 21 Jul '13

21 Jul '13

On 07/20/2013 11:25 AM, Jean-Francois Moine wrote: > This generic simple card driver uses DT values to instanciate an audio > system in which the real work is done by the subdrivers (audio > controller, audio codec and pcm/dma controller). New DT bindings should be sent to devicetree(a)vger.kernel.org for review. Note that's a branch new list (it moved from a different server), so it might be best to wait a few days for people to subscribe before sending mail to it.

1 0

[alsa-devel] [PATCH RFC 00/10] Refactor set_format()
by Eldad Zack 20 Jul '13

20 Jul '13

Hi, This is a low-priority series (applies against current for-next). I had the idea of refactoring set_format to make it easier to read (I actually indended to look at the implicit feedback setup race, but stumbled on this again). This series is not intended to be applied as-is (it has some whitespace issues and I wrote the commit messages quickly), but to get some feedback - if you think it's a good idea in general, and also specific feedback to individual patches. Assuming it makes sense, I'll fix the whitespace issues, commit messages and anything else before resubmitting. Cheers, Eldad Eldad Zack (10): ALSA: usb-audio: remove disabled debug code in set_format ALSA: usb-audio: remove assignment from if condition ALSA: usb-audio: separate sync endpoint setting from set_format ALSA: usb-audio: move implicit fb quirks to separate function ALSA: usb-audio: reverse condition logic in set_sync_endpoint ALSA: usb-audio: do not initialize and check implicit_fb ALSA: usb-audio: remove is_playback from implicit feedback quirks ALSA: usb-audio: remove implicit_fb from quirk ALSA: usb-audio: clarify implicit fb quirk for Roland ALSA: usb-audio: check alts is not NULL before adding ep sound/usb/pcm.c | 249 ++++++++++++++++++++++++++++++++------------------------ 1 file changed, 144 insertions(+), 105 deletions(-) -- 1.8.1.5

2 12

Re: [alsa-devel] [ecasound] Garbage at end of playback: ecasound or alsa broken?
by Knut Petersen 20 Jul '13

20 Jul '13

On 20.07.2013 11:57, Julien Claassen wrote: > Hello Knut! > No idea yet, who's to blame. but have you tried repeating the experiment > with another tool. You can use audacity, so you can use the GUI. Audacity is even more broken. That´s true for the version from opensuse (2.03), and it´s true for the current development code, If I mark that 1s of audio described in the original post, a lot of audio is missing at the beginning and a bit of audio is missing at end. In total only about 460 msec of audio is captured, 540 msec is lost. Audacity currently does not playback correctly with non-memory-mapped alsa devices. But that´s another story ;-( > There are other recording applications. Or you might even try with the alsa utilities > aplay and arecord. this will require two console or terminals. If I put arecord -f cd -D hw:0,0 -d 2 testarec.wav & aplay testin.wav -D hw:0,0 into a script I get a perfect recording: 47 msec silence, exactly 1s of audio data, silence up to the end of the file. Well, that result seems to indicate that I found an ecasound bug. Unless it´s some alsa code unused by aplay ;-) > You might also try the same experiment with JACK, if you're not averse to it. In the best case that could only mask the bug. > I did digitise a few tapes and prelistened to the volume and audio quality. > So it's not a problem in general. > Half a second silence at the end of the audio masks the bug. Because of that I agree that it´s not a general problem, most people will never notice it. But it´s still a bug. cu, Knut

1 0

[alsa-devel] [PATCH] ASoC: twl6040: Use power efficient workqueue
by Mark Brown 20 Jul '13

20 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621dd..caf8784 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.8.3.2

3 2

[alsa-devel] [PATCH v2] compress: fix the typo in compress_get_alsa_rate()
by Vinod Koul 20 Jul '13

20 Jul '13

Reported-by: Ravi <ralama(a)codeaurora.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- utils.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/utils.c b/utils.c index fa37f23..a835bef 100644 --- a/utils.c +++ b/utils.c @@ -72,7 +72,7 @@ unsigned int compress_get_alsa_rate(unsigned int rate) return SNDRV_PCM_RATE_11025; case 16000: return SNDRV_PCM_RATE_16000; - case 220500: + case 22050: return SNDRV_PCM_RATE_22050; case 32000: return SNDRV_PCM_RATE_32000; -- 1.7.0.4

1 0

[alsa-devel] [PATCH] compress: fix the typo is get rates
by Vinod Koul 20 Jul '13

20 Jul '13

Reported-by: Ravi <ralama(a)codeaurora.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- utils.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/utils.c b/utils.c index fa37f23..a835bef 100644 --- a/utils.c +++ b/utils.c @@ -72,7 +72,7 @@ unsigned int compress_get_alsa_rate(unsigned int rate) return SNDRV_PCM_RATE_11025; case 16000: return SNDRV_PCM_RATE_16000; - case 220500: + case 22050: return SNDRV_PCM_RATE_22050; case 32000: return SNDRV_PCM_RATE_32000; -- 1.7.0.4

1 0

Re: [alsa-devel] [PATCH] ASoC: ep93xx: fix build of ep93xx-ac97.c
by Mark Brown 19 Jul '13

19 Jul '13

On Fri, Jul 19, 2013 at 09:53:25AM -0700, H Hartley Sweeten wrote: > Fix the build of this driver. It was broken by: > > Commit 453807f3006757a5661c4000262d7d9284b5214c > ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: atmel: add wm8904 based audio machine driver
by Bo Shen 19 Jul '13

19 Jul '13

Add wm8904 based audio machine driver for Atmel EK board The following ek board based on it - at91sam9n12ek - sama5d3xek (d31, d33, d34, d35) Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- This patch is only for Atmel SoC with wm8904 codec machine driver only, temp no change for dts file, as there are many changes to dts file in mailing list for atmel SoC now, so hold on for dts change, after that will send the dts file patches Change in v2: - Update the binding doc, add widgets describe of codec and board - set sysclk after set pll - using bias level trasition to decide enable/disable clock - return real error code instead of return -1 Todo: - After the at91 clock move to common clock framework, then rework the clock in this driver --- .../devicetree/bindings/sound/atmel-wm8904.txt | 55 +++++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_wm8904.c | 254 ++++++++++++++++++++ 4 files changed, 321 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 sound/soc/atmel/atmel_wm8904.c diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt new file mode 100644 index 0000000..8bbe50c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt @@ -0,0 +1,55 @@ +Atmel ASoC driver with wm8904 audio codec complex + +Required properties: + - compatible: "atmel,asoc-wm8904" + - atmel,model: The user-visible name of this sound complex. + - atmel,audio-routing: A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the WM8904's pins, and the jacks on the board: + + WM8904 pins: + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + + Board connectors: + + * Headphone Jack + * Line In Jack + * Mic + + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8904 audio codec + +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; +}; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1c0b185..986323b 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -33,6 +33,16 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967cc..922d4da 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,8 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o +snd-atmel-soc-wm8904-objs := atmel_wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c new file mode 100644 index 0000000..7222380 --- /dev/null +++ b/sound/soc/atmel/atmel_wm8904.c @@ -0,0 +1,254 @@ +/* + * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec. + * + * Copyright (C) 2012 Atmel + * + * Author: Bo Shen <voice.shen(a)atmel.com> + * + * GPLv2 or later + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/pinctrl/consumer.h> + +#include <sound/soc.h> + +#include "../codecs/wm8904.h" +#include "atmel_ssc_dai.h" + +#define MCLK_RATE 32768 + +static struct clk *mclk; + +static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, + 32768, params_rate(params) * 256); + if (ret < 0) { + pr_err("%s - failed to set wm8904 codec PLL.", __func__); + return ret; + } + + /* + * As here wm8904 use FLL output as its system clock + * so calling set_sysclk won't care freq parameter + * then we pass 0 + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("%s -failed to set wm8904 SYSCLK\n", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops atmel_asoc_wm8904_ops = { + .hw_params = atmel_asoc_wm8904_hw_params, +}; + +static int atmel_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + switch (level) { + case SND_SOC_BIAS_PREPARE: + clk_prepare_enable(mclk); + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(mclk); + break; + default: + break; + } + } + + return 0; +}; + +static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { + .name = "WM8904", + .stream_name = "WM8904 PCM", + .codec_dai_name = "wm8904-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &atmel_asoc_wm8904_ops, +}; + +static struct snd_soc_card atmel_asoc_wm8904_card = { + .name = "atmel_asoc_wm8904", + .owner = THIS_MODULE, + .set_bias_level = atmel_set_bias_level, + .dai_link = &atmel_asoc_wm8904_dailink, + .num_links = 1, + .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets), + .fully_routed = true, +}; + +static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int ret; + + if (!np) { + dev_err(&pdev->dev, "only device tree supported\n"); + return -EINVAL; + } + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "failed to parse card name\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio routing\n"); + return ret; + } + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "failed to get dai and pcm info\n"); + ret = -EINVAL; + return ret; + } + dailink->cpu_of_node = cpu_np; + dailink->platform_of_node = cpu_np; + of_node_put(cpu_np); + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "failed to get codec info\n"); + ret = -EINVAL; + return ret; + } + dailink->codec_of_node = codec_np; + of_node_put(codec_np); + + return 0; +} + +static int atmel_asoc_wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + struct clk *clk_src; + struct pinctrl *pinctrl; + int id, ret; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } + + card->dev = &pdev->dev; + ret = atmel_asoc_wm8904_dt_init(pdev); + if (ret) { + dev_err(&pdev->dev, "failed to init dt info\n"); + return ret; + } + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + ret = atmel_ssc_set_audio(id); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id); + return ret; + } + + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "failed to get pck0\n"); + ret = PTR_ERR(mclk); + goto err_set_audio; + } + + clk_src = clk_get(NULL, "clk32k"); + if (IS_ERR(clk_src)) { + dev_err(&pdev->dev, "failed to get clk32k\n"); + ret = PTR_ERR(clk_src); + goto err_set_audio; + } + + ret = clk_set_parent(mclk, clk_src); + clk_put(clk_src); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set MCLK parent\n"); + goto err_set_audio; + } + + dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); + clk_set_rate(mclk, MCLK_RATE); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + goto err_set_audio; + } + + return 0; + +err_set_audio: + atmel_ssc_put_audio(id); + return ret; +} + +static int atmel_asoc_wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int id; + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(id); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { + { .compatible = "atmel,asoc-wm8904", }, + { } +}; +#endif + +static struct platform_driver atmel_asoc_wm8904_driver = { + .driver = { + .name = "atmel-wm8904-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + }, + .probe = atmel_asoc_wm8904_probe, + .remove = atmel_asoc_wm8904_remove, +}; + +module_platform_driver(atmel_asoc_wm8904_driver); + +/* Module information */ +MODULE_AUTHOR("Bo Shen <voice.shen(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: wm8962: Do not call configure_bclk() inside wm8962_set_dai_sysclk()
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. Quoting Mark Brown [1]: "The issue here is triggered by the machine switching from the FLL to direct MCLK usage where the MCLK isn't generating a useful ratio. I suspect we should just kill the configure_bclk() in set_sysclk(), that one isn't safe as we can't reconfigure a live SYSCLK and it's probably the one that generates your warnings." Confirmed that the "Unsupported sysclk ratio" error message comes from wm8962_set_dai_sysclk(), so get rid of wm8962_configure_bclk() inside this function. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064241.html Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Remove wm8962_configure_bclk() from wm8962_set_dai_sysclk sound/soc/codecs/wm8962.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ec..8b8905d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2621,8 +2621,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/13] Remove static quirks for HD-audio AD codecs
by Takashi Iwai 19 Jul '13

19 Jul '13

OK, it's time to clean up the craps in the old room. Let's start from AD codecs, which I could test a few of them, and worked well with the generic parser. [PATCH 01/13] ALSA: hda - Add GPIO control to AD1884 HP fixup [PATCH 02/13] ALSA: hda - Add fixup for Lenovo Thinkpad with AD1984 [PATCH 03/13] ALSA: hda - Add fixup for HP TouchSmart with AD1984A [PATCH 04/13] ALSA: hda - Remove static quirks for AD1882 [PATCH 05/13] ALSA: hda - Remove static quirks for AD1884/1984 & [PATCH 06/13] ALSA: hda - Remove static quirks for AD1981 and AD1983 [PATCH 07/13] ALSA: hda - Remove static quirks for AD1988 codecs [PATCH 08/13] ALSA: hda - Convert the static quirk for Samsung Q1 [PATCH 09/13] ALSA: hda - Convert static quirks for AD1986A Samsung [PATCH 10/13] ALSA: hda - Drop static quirks for other AD1986A [PATCH 11/13] ALSA: hda - Drop static quirk for Toshiba Satellite [PATCH 12/13] ALSA: hda - Drop a few other static quirks for AD1986A [PATCH 13/13] ALSA: hda - Remove static quirks for AD1986A codec Takashi

1 13

[alsa-devel] [RFC][PATCH 0/4] firewire-lib: add payload callback function for each driver
by Takashi Sakamoto 19 Jul '13

19 Jul '13

Hi Clemens, (CCed Rober von Chamier and Damien Zammit, they work for Dice and 003Rack driver) Would you please review my patches and comment? Currently snd-firewire-lib handles the contents of data block for AMDTP packet in itself. But to implement drivers for Fireworks(Echo Audio)/BeBoB(BridgeCo), there is an inconvinience. Fireworks don't use AM824 label for PCM data. BeBoB needs to remap PCM and MIDI channels depending on devices. And Dice(TC Applied Technologies) - I think you understand this than me - needs the mode called as "dual wire". In this mode, the sampling rate is a half of requested one and each data block includes the data as twice as usual sampling rate. The each data block handle two PCM frames, usually it's one. Your snd-firewire-lib in alsa-kprivate.git repository manages to solve these issue. But it has restriction about the PCM format (S32 only) and for me it seems to be hard to maintain. Additionally to implement full duplex with synchronization, I need to restart streams for MIDI. Then current amdtp_out_stream_set_pcm_format() is not convinient. I want to add condition for PCM format into each drivers. Furthermore, 003Rack(Digidesign) - I don't work for this - has more issue. The differences related to payload are: - the value of syt field in AMDTP packet is always zero - The data in each channel is encoded by its own way The driver needs to refer to and modify the contents of payload. To solve these device-dependent issues, I want to move the processing of contents in payload from firewire-lib to each devices. This series of patches is for this purpose. sakamocchi (4): firewire-lib: add helper function to set data block size firewire-lib: add callback function for payload firewire-lib: remove unused members and functions firewire-speakers: add functions to handle payload amdtp.c | 144 ++++++------------------------------------------------------ amdtp.h | 45 +++++++------------ speakers.c | 94 ++++++++++++++++++++++++++++++++++++--- 3 files changed, 117 insertions(+), 166 deletions(-) -- 1.7.10.4

2 6

[alsa-devel] [PATCH] ASoC: wm8962: Provide a .hw_free function
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. In order to avoid such error message, let's add a .hw_free function which will zero out wm8962->lrclk and wm8962->bclk, which according to the code below: if (!wm8962->bclk || !wm8962->lrclk) { dev_dbg(codec->dev, "No audio clocks configured\n"); return; } ,will cause the wm8962_configure_bclk() function not to search a valid sysclk ratio after playback has finished. Suggested-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8962.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6a5066c..01033c2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,6 +2914,18 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute) WM8962_DAC_MUTE, val); } +static int wm8962_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962->lrclk = 0; + wm8962->bclk = 0; + + return 0; +} + #define WM8962_RATES SNDRV_PCM_RATE_8000_96000 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -2921,6 +2933,7 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute) static const struct snd_soc_dai_ops wm8962_dai_ops = { .hw_params = wm8962_hw_params, + .hw_free = wm8962_hw_free, .set_sysclk = wm8962_set_dai_sysclk, .set_fmt = wm8962_set_dai_fmt, .digital_mute = wm8962_mute, -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Provide register I/O functions by default
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> Use the ARM version by default as that's the more generally portable one, it doesn't matter if they work well on random platforms when the goal is only build coverage. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c9974a4..e12a997 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -36,7 +36,7 @@ #define read_ssi(addr) in_be32(addr) #define write_ssi(val, addr) out_be32(addr, val) #define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) -#elif defined ARM +#else #define read_ssi(addr) readl(addr) #define write_ssi(val, addr) writel(val, addr) /* -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: atmel: add wm8904 based audio machine driver
by Bo Shen 19 Jul '13

19 Jul '13

add wm8904 based audio machine driver the following ek board based on it - at91sam9n12ek - sama5d3xek (d31, d33, d34, d35) Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- This patch is only for Atmel SoC with wm8904 codec machine driver only, temp no change for dts file, as there are many changes to dts file in mailing list for atmel SoC now, so hold on for dts change, after that will send the dts file patches --- .../devicetree/bindings/sound/atmel-wm8904.txt | 30 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_wm8904.c | 263 ++++++++++++++++++++ 4 files changed, 305 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 sound/soc/atmel/atmel_wm8904.c diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt new file mode 100644 index 0000000..5990271 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt @@ -0,0 +1,30 @@ +Atmel ASoC driver with wm8904 audio codec complex + +Required properties: + - compatible: "atmel,asoc-wm8904" + - atmel,model: The user-visible name of this sound complex. + - atmel,audio-routing: A list of the connections between audio components. + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8904 audio codec +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; +}; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1c0b185..986323b 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -33,6 +33,16 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967cc..922d4da 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,8 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o +snd-atmel-soc-wm8904-objs := atmel_wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c new file mode 100644 index 0000000..e309607 --- /dev/null +++ b/sound/soc/atmel/atmel_wm8904.c @@ -0,0 +1,263 @@ +/* + * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec. + * + * Copyright (C) 2012 Atmel + * + * Author: Bo Shen <voice.shen(a)atmel.com> + * + * GPLv2 or later + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/pinctrl/consumer.h> + +#include <sound/soc.h> + +#include "../codecs/wm8904.h" +#include "atmel_ssc_dai.h" + +#define MCLK_RATE 32768 + +static struct clk *mclk; + +static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), +}; + +static int atmel_asoc_wm8904_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("%s -failed to set wm8904 SYSCLK\n", __func__); + return ret; + } + + return 0; +} + +static int atmel_asoc_wm8904_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, + 32768, params_rate(params) * 256); + if (ret < 0) { + pr_err("%s - failed to set wm8904 codec PLL.", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops atmel_asoc_wm8904_ops = { + .hw_params = atmel_asoc_wm8904_params, +}; + +static int atmel_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + static int mclk_on; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!mclk_on) { + ret = clk_prepare_enable(mclk); + if (ret == 0) + mclk_on = 1; + } + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + if (mclk_on) + clk_disable_unprepare(mclk); + mclk_on = 0; + break; + } + + return ret; +}; + +static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { + .name = "WM8904", + .stream_name = "WM8904 PCM", + .codec_dai_name = "wm8904-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = atmel_asoc_wm8904_init, + .ops = &atmel_asoc_wm8904_ops, +}; + +static struct snd_soc_card atmel_asoc_wm8904_card = { + .name = "atmel_asoc_wm8904", + .owner = THIS_MODULE, + .set_bias_level = atmel_set_bias_level, + .dai_link = &atmel_asoc_wm8904_dailink, + .num_links = 1, + .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets), + .fully_routed = true, +}; + +static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int ret; + + if (!np) + return -1; + + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) { + dev_err(&pdev->dev, "failed to parse card name\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio routing\n"); + return ret; + } + + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "failed to get dai and pcm info\n"); + ret = -EINVAL; + return ret; + } + dailink->cpu_of_node = cpu_np; + dailink->platform_of_node = cpu_np; + of_node_put(cpu_np); + + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "failed to get codec info\n"); + ret = -EINVAL; + return ret; + } + dailink->codec_of_node = codec_np; + of_node_put(codec_np); + + return 0; +} + +static int atmel_asoc_wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &atmel_asoc_wm8904_card; + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + struct clk *clk_src; + struct pinctrl *pinctrl; + int id, ret; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } + + card->dev = &pdev->dev; + ret = atmel_asoc_wm8904_dt_init(pdev); + if (ret) { + dev_err(&pdev->dev, "failed to init dt info\n"); + goto err_set_audio; + } + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + ret = atmel_ssc_set_audio(id); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set SSC 0 for audio\n"); + return ret; + } + + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "failed to get pck0\n"); + ret = PTR_ERR(mclk); + goto err_set_audio; + } + + clk_src = clk_get(NULL, "clk32k"); + if (IS_ERR(clk_src)) { + dev_err(&pdev->dev, "failed to get clk32k\n"); + ret = PTR_ERR(clk_src); + goto err_set_audio; + } + + ret = clk_set_parent(mclk, clk_src); + clk_put(clk_src); + if (ret != 0) { + dev_err(&pdev->dev, "failed to set MCLK parent\n"); + goto err_set_audio; + } + + dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); + clk_set_rate(mclk, MCLK_RATE); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + goto err_set_audio; + } + + return 0; + +err_set_audio: + atmel_ssc_put_audio(0); + return ret; +} + +static int atmel_asoc_wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; + int id; + + id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc"); + + snd_soc_unregister_card(card); + atmel_ssc_put_audio(id); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { + { .compatible = "atmel,asoc-wm8904", }, + { } +}; +#endif + +static struct platform_driver atmel_asoc_wm8904_driver = { + .driver = { + .name = "atmel-wm8904-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + }, + .probe = atmel_asoc_wm8904_probe, + .remove = atmel_asoc_wm8904_remove, +}; + +module_platform_driver(atmel_asoc_wm8904_driver); + +/* Module information */ +MODULE_AUTHOR("Bo Shen <voice.shen(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: wm8962: No need to call wm8962_configure_bclk() multiple times
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. There is no need to re-calculate the bit clock after the playback has finished, so call wm8962_configure_bclk only once inside wm8962_hw_params(). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8962.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6a5066c..1bf79dc 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2497,8 +2497,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*50k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x80); - - wm8962_configure_bclk(codec); break; case SND_SOC_BIAS_STANDBY: @@ -2621,8 +2619,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } -- 1.8.1.2

3 2

[alsa-devel] [PATCH] hda-emu: Allow Realtek codecs to access node 0x51
by David Henningsson 19 Jul '13

19 Jul '13

There is sometimes a secret node 0x51 on Realtek codecs, controlling an equally secret DSP. At least make sure the automated test suite doesn't fail when trying to access this secret node. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 6 ++++++ hda-spec.c | 13 +++++++++++++ include/hda-types.h | 1 + 3 files changed, 20 insertions(+) diff --git a/hda-int.c b/hda-int.c index fbd3cee..ef90fe8 100644 --- a/hda-int.c +++ b/hda-int.c @@ -1125,6 +1125,12 @@ int hda_cmd(struct xhda_codec *codec, unsigned int cmd) unsigned int nid = (cmd >> 20) & 0x7f; unsigned int verb = (cmd >> 8) & 0xfff; + if (codec->extended_cmd) { + int r = codec->extended_cmd(codec, cmd); + if (r != -ENXIO) + return r; + } + tbl = find_matching_verb(codec, verb); if (!tbl) return -ENXIO; diff --git a/hda-spec.c b/hda-spec.c index 8f95914..e4bb547 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -354,6 +354,16 @@ static struct verb_ext_list extensions[] = { { } }; +static int realtek_dsp_cmd(struct xhda_codec *codec, unsigned int cmd) +{ + unsigned int nid = (cmd >> 20) & 0x7f; + if (nid != 0x51) + return -ENXIO; + /* There might be a secret DSP connected to node 0x51 */ + codec->rc = 0; + return 0; +} + /* * fixups */ @@ -466,4 +476,7 @@ void add_codec_extensions(struct xhda_codec *codec) else if (!strcmp(codec->parsed_name, "Realtek ALC661")) hda_set_proc_coef(codec, 0x20, 0x00, 0x8020); + if (!strncmp(codec->parsed_name, "Realtek", strlen("Realtek"))) + codec->extended_cmd = realtek_dsp_cmd; + } diff --git a/include/hda-types.h b/include/hda-types.h index fac5084..fe6b31f 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -114,6 +114,7 @@ struct xhda_codec { unsigned int rc; struct xhda_verb_table *extended_verbs; struct xhda_verb_table *extended_parameters; + int (*extended_cmd) (struct xhda_codec *codec, unsigned int cmd); struct xhda_value_cache *value_cache; unsigned short pci_vendor, pci_device; unsigned short pci_subvendor, pci_subdevice; -- 1.7.9.5

2 1

[alsa-devel] [RFC] ASoC: wm892: Set an initial non-mute volume for the headphones
by Fabio Estevam 19 Jul '13

19 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> By default the headphone left/right registers have a volume of zero. Instead of having to change the volume via alsamixer in order to be able to hear something, let's set an initial value of -20dB, so that audio playback can be heard by default. Tested on a mx6qsabresd board. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Mark, I posted it as RFC as I don't know if this is a valid approach. Suggestions are welcome. Thanks sound/soc/codecs/wm8962.c | 6 ++++++ sound/soc/codecs/wm8962.h | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ec..6a5066c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3448,6 +3448,12 @@ static int wm8962_probe(struct snd_soc_codec *codec) WM8962_MICBIAS_LVL, pdata->mic_cfg); + /* set an initial volume for playback of -20 dB */ + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUTL_VOL_MASK, WM8962_HPOUTx_MINUS_20_DB); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUTR_VOL_MASK, WM8962_HPOUTx_MINUS_20_DB); + /* Latch volume update bits */ snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, WM8962_IN_VU, WM8962_IN_VU); diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index a1a5d52..5dbe972 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -27,6 +27,8 @@ #define WM8962_FLL_OSC 3 #define WM8962_FLL_INT 4 +#define WM8962_HPOUTx_MINUS_20_DB 0x65 + /* * Register values. */ -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: jack: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..7aa26b5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; -- 1.8.3.2

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm8350: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8350.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0e8b3aa..af1318d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,8 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1319,8 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.8.3.2

2 4

[alsa-devel] [PATCH] ASoC: sta32x: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sta32x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index cfb55fe..06edb39 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -363,16 +363,18 @@ static void sta32x_watchdog(struct work_struct *work) } if (!sta32x->shutdown) - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } static void sta32x_watchdog_start(struct sta32x_priv *sta32x) { if (sta32x->pdata->needs_esd_watchdog) { sta32x->shutdown = 0; - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } } -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: max98090: Use power efficient workqueue
by Mark Brown 19 Jul '13

19 Jul '13

From: Mark Brown <broonie(a)linaro.org> None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98090.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ad5313f..0569a4c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2084,8 +2084,9 @@ static irqreturn_t max98090_interrupt(int irq, void *data) pm_wakeup_event(codec->dev, 100); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); } if (active & M98090_DRCACT_MASK) @@ -2132,8 +2133,9 @@ int max98090_mic_detect(struct snd_soc_codec *codec, snd_soc_jack_report(max98090->jack, 0, SND_JACK_HEADSET | SND_JACK_BTN_0); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); return 0; } -- 1.8.3.2

2 1

[alsa-devel] cs42l52.c:undefined reference to `devm_input_allocate_device'
by Fengguang Wu 19 Jul '13

19 Jul '13

Greetings, Here is an old link error, unveiled by this build fix: tree: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master head: 7a62711aacda8887d94c40daa199b37abb1d54e1 commit: 446d64e3e1154806092ac27de198dff1225797d9 block: fix part_pack_uuid() build error date: 5 months ago config: i386-randconfig-x002-0719 (attached as .config) All error/warnings: sound/built-in.o: In function `cs42l52_beep_set': cs42l52.c:(.text+0x425b3): undefined reference to `input_event' sound/built-in.o: In function `cs42l52_probe': >> cs42l52.c:(.text+0x42776): undefined reference to `devm_input_allocate_device' cs42l52.c:(.text+0x42869): undefined reference to `input_register_device' sound/built-in.o: In function `wm8962_beep_set': wm8962.c:(.text+0x7733b): undefined reference to `input_event' sound/built-in.o: In function `wm8962_probe': >> wm8962.c:(.text+0x786c5): undefined reference to `devm_input_allocate_device' wm8962.c:(.text+0x787ea): undefined reference to `input_register_device' --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation # # Automatically generated file; DO NOT EDIT. # Linux/i386 3.8.0 Kernel Configuration # # CONFIG_64BIT is not set CONFIG_X86_32=y CONFIG_X86=y CONFIG_INSTRUCTION_DECODER=y CONFIG_OUTPUT_FORMAT="elf32-i386" CONFIG_ARCH_DEFCONFIG="arch/x86/configs/i386_defconfig" CONFIG_LOCKDEP_SUPPORT=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_HAVE_LATENCYTOP_SUPPORT=y CONFIG_MMU=y CONFIG_NEED_SG_DMA_LENGTH=y CONFIG_GENERIC_ISA_DMA=y CONFIG_GENERIC_BUG=y CONFIG_GENERIC_HWEIGHT=y CONFIG_GENERIC_GPIO=y CONFIG_ARCH_MAY_HAVE_PC_FDC=y CONFIG_RWSEM_XCHGADD_ALGORITHM=y CONFIG_GENERIC_CALIBRATE_DELAY=y CONFIG_ARCH_HAS_CPU_RELAX=y CONFIG_ARCH_HAS_DEFAULT_IDLE=y CONFIG_ARCH_HAS_CACHE_LINE_SIZE=y CONFIG_ARCH_HAS_CPU_AUTOPROBE=y CONFIG_HAVE_SETUP_PER_CPU_AREA=y CONFIG_NEED_PER_CPU_EMBED_FIRST_CHUNK=y CONFIG_NEED_PER_CPU_PAGE_FIRST_CHUNK=y CONFIG_ARCH_HIBERNATION_POSSIBLE=y CONFIG_ARCH_SUSPEND_POSSIBLE=y # CONFIG_ZONE_DMA32 is not set # CONFIG_AUDIT_ARCH is not set CONFIG_ARCH_SUPPORTS_OPTIMIZED_INLINING=y CONFIG_ARCH_SUPPORTS_DEBUG_PAGEALLOC=y CONFIG_X86_32_LAZY_GS=y CONFIG_ARCH_HWEIGHT_CFLAGS="-fcall-saved-ecx -fcall-saved-edx" CONFIG_ARCH_SUPPORTS_UPROBES=y CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" CONFIG_CONSTRUCTORS=y CONFIG_IRQ_WORK=y CONFIG_BUILDTIME_EXTABLE_SORT=y # # General setup # CONFIG_EXPERIMENTAL=y CONFIG_BROKEN_ON_SMP=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_CROSS_COMPILE="" CONFIG_LOCALVERSION="" CONFIG_LOCALVERSION_AUTO=y CONFIG_HAVE_KERNEL_GZIP=y CONFIG_HAVE_KERNEL_BZIP2=y CONFIG_HAVE_KERNEL_LZMA=y CONFIG_HAVE_KERNEL_XZ=y CONFIG_HAVE_KERNEL_LZO=y # CONFIG_KERNEL_GZIP is not set CONFIG_KERNEL_BZIP2=y # CONFIG_KERNEL_LZMA is not set # CONFIG_KERNEL_XZ is not set # CONFIG_KERNEL_LZO is not set CONFIG_DEFAULT_HOSTNAME="(none)" # CONFIG_SYSVIPC is not set # CONFIG_FHANDLE is not set CONFIG_HAVE_GENERIC_HARDIRQS=y # # IRQ subsystem # CONFIG_GENERIC_HARDIRQS=y CONFIG_GENERIC_IRQ_PROBE=y CONFIG_GENERIC_IRQ_SHOW=y CONFIG_GENERIC_IRQ_CHIP=y CONFIG_IRQ_DOMAIN=y # CONFIG_IRQ_DOMAIN_DEBUG is not set CONFIG_IRQ_FORCED_THREADING=y CONFIG_SPARSE_IRQ=y CONFIG_CLOCKSOURCE_WATCHDOG=y CONFIG_ALWAYS_USE_PERSISTENT_CLOCK=y CONFIG_KTIME_SCALAR=y CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y CONFIG_GENERIC_CLOCKEVENTS_MIN_ADJUST=y CONFIG_GENERIC_CMOS_UPDATE=y # # Timers subsystem # CONFIG_TICK_ONESHOT=y # CONFIG_NO_HZ is not set CONFIG_HIGH_RES_TIMERS=y # # CPU/Task time and stats accounting # CONFIG_TICK_CPU_ACCOUNTING=y # CONFIG_IRQ_TIME_ACCOUNTING is not set # CONFIG_BSD_PROCESS_ACCT is not set # # RCU Subsystem # CONFIG_TREE_PREEMPT_RCU=y # CONFIG_TINY_PREEMPT_RCU is not set CONFIG_PREEMPT_RCU=y CONFIG_RCU_STALL_COMMON=y CONFIG_RCU_FANOUT=32 CONFIG_RCU_FANOUT_LEAF=16 # CONFIG_RCU_FANOUT_EXACT is not set # CONFIG_TREE_RCU_TRACE is not set # CONFIG_RCU_BOOST is not set # CONFIG_RCU_NOCB_CPU is not set # CONFIG_IKCONFIG is not set CONFIG_LOG_BUF_SHIFT=17 CONFIG_HAVE_UNSTABLE_SCHED_CLOCK=y CONFIG_ARCH_SUPPORTS_NUMA_BALANCING=y CONFIG_ARCH_WANTS_PROT_NUMA_PROT_NONE=y # CONFIG_CHECKPOINT_RESTORE is not set CONFIG_NAMESPACES=y # CONFIG_UTS_NS is not set CONFIG_USER_NS=y CONFIG_PID_NS=y CONFIG_UIDGID_CONVERTED=y CONFIG_UIDGID_STRICT_TYPE_CHECKS=y # CONFIG_SCHED_AUTOGROUP is not set CONFIG_SYSFS_DEPRECATED=y CONFIG_SYSFS_DEPRECATED_V2=y # CONFIG_RELAY is not set # CONFIG_BLK_DEV_INITRD is not set CONFIG_CC_OPTIMIZE_FOR_SIZE=y CONFIG_ANON_INODES=y CONFIG_EXPERT=y CONFIG_HAVE_UID16=y # CONFIG_UID16 is not set CONFIG_SYSCTL_EXCEPTION_TRACE=y CONFIG_KALLSYMS=y CONFIG_KALLSYMS_ALL=y CONFIG_HOTPLUG=y CONFIG_PRINTK=y CONFIG_BUG=y CONFIG_ELF_CORE=y # CONFIG_PCSPKR_PLATFORM is not set CONFIG_HAVE_PCSPKR_PLATFORM=y # CONFIG_BASE_FULL is not set # CONFIG_FUTEX is not set CONFIG_EPOLL=y # CONFIG_SIGNALFD is not set # CONFIG_TIMERFD is not set # CONFIG_EVENTFD is not set CONFIG_SHMEM=y CONFIG_AIO=y CONFIG_EMBEDDED=y CONFIG_HAVE_PERF_EVENTS=y # # Kernel Performance Events And Counters # CONFIG_PERF_EVENTS=y # CONFIG_DEBUG_PERF_USE_VMALLOC is not set CONFIG_VM_EVENT_COUNTERS=y # CONFIG_PCI_QUIRKS is not set CONFIG_COMPAT_BRK=y CONFIG_SLAB=y # CONFIG_SLUB is not set # CONFIG_SLOB is not set CONFIG_PROFILING=y CONFIG_TRACEPOINTS=y # CONFIG_OPROFILE is not set CONFIG_HAVE_OPROFILE=y CONFIG_OPROFILE_NMI_TIMER=y # CONFIG_KPROBES is not set # CONFIG_JUMP_LABEL is not set CONFIG_UPROBES=y CONFIG_HAVE_EFFICIENT_UNALIGNED_ACCESS=y CONFIG_ARCH_USE_BUILTIN_BSWAP=y CONFIG_HAVE_IOREMAP_PROT=y CONFIG_HAVE_KPROBES=y CONFIG_HAVE_KRETPROBES=y CONFIG_HAVE_OPTPROBES=y CONFIG_HAVE_KPROBES_ON_FTRACE=y CONFIG_HAVE_ARCH_TRACEHOOK=y CONFIG_HAVE_DMA_ATTRS=y CONFIG_HAVE_DMA_CONTIGUOUS=y CONFIG_GENERIC_SMP_IDLE_THREAD=y CONFIG_HAVE_REGS_AND_STACK_ACCESS_API=y CONFIG_HAVE_DMA_API_DEBUG=y CONFIG_HAVE_HW_BREAKPOINT=y CONFIG_HAVE_MIXED_BREAKPOINTS_REGS=y CONFIG_HAVE_USER_RETURN_NOTIFIER=y CONFIG_HAVE_PERF_EVENTS_NMI=y CONFIG_HAVE_PERF_REGS=y CONFIG_HAVE_PERF_USER_STACK_DUMP=y CONFIG_HAVE_ARCH_JUMP_LABEL=y CONFIG_ARCH_HAVE_NMI_SAFE_CMPXCHG=y CONFIG_HAVE_CMPXCHG_LOCAL=y CONFIG_HAVE_CMPXCHG_DOUBLE=y CONFIG_ARCH_WANT_IPC_PARSE_VERSION=y CONFIG_HAVE_ARCH_SECCOMP_FILTER=y CONFIG_HAVE_IRQ_TIME_ACCOUNTING=y CONFIG_HAVE_ARCH_TRANSPARENT_HUGEPAGE=y CONFIG_MODULES_USE_ELF_REL=y CONFIG_CLONE_BACKWARDS=y CONFIG_OLD_SIGSUSPEND3=y CONFIG_OLD_SIGACTION=y # # GCOV-based kernel profiling # CONFIG_GCOV_KERNEL=y # CONFIG_GCOV_PROFILE_ALL is not set CONFIG_HAVE_GENERIC_DMA_COHERENT=y CONFIG_RT_MUTEXES=y CONFIG_BASE_SMALL=1 CONFIG_MODULES=y CONFIG_MODULE_FORCE_LOAD=y CONFIG_MODULE_UNLOAD=y # CONFIG_MODULE_FORCE_UNLOAD is not set CONFIG_MODVERSIONS=y CONFIG_MODULE_SRCVERSION_ALL=y CONFIG_MODULE_SIG=y # CONFIG_MODULE_SIG_FORCE is not set # CONFIG_MODULE_SIG_SHA1 is not set # CONFIG_MODULE_SIG_SHA224 is not set # CONFIG_MODULE_SIG_SHA256 is not set CONFIG_MODULE_SIG_SHA384=y # CONFIG_MODULE_SIG_SHA512 is not set # CONFIG_BLOCK is not set CONFIG_ASN1=y CONFIG_UNINLINE_SPIN_UNLOCK=y # CONFIG_FREEZER is not set # # Processor type and features # CONFIG_ZONE_DMA=y # CONFIG_SMP is not set CONFIG_X86_MPPARSE=y # CONFIG_X86_EXTENDED_PLATFORM is not set # CONFIG_X86_GOLDFISH is not set CONFIG_X86_SUPPORTS_MEMORY_FAILURE=y CONFIG_X86_32_IRIS=y CONFIG_SCHED_OMIT_FRAME_POINTER=y # CONFIG_PARAVIRT_GUEST is not set CONFIG_NO_BOOTMEM=y # CONFIG_MEMTEST is not set # CONFIG_M486 is not set # CONFIG_M586 is not set # CONFIG_M586TSC is not set # CONFIG_M586MMX is not set # CONFIG_M686 is not set # CONFIG_MPENTIUMII is not set # CONFIG_MPENTIUMIII is not set # CONFIG_MPENTIUMM is not set # CONFIG_MPENTIUM4 is not set # CONFIG_MK6 is not set # CONFIG_MK7 is not set # CONFIG_MK8 is not set # CONFIG_MCRUSOE is not set # CONFIG_MEFFICEON is not set # CONFIG_MWINCHIPC6 is not set # CONFIG_MWINCHIP3D is not set # CONFIG_MELAN is not set # CONFIG_MGEODEGX1 is not set CONFIG_MGEODE_LX=y # CONFIG_MCYRIXIII is not set # CONFIG_MVIAC3_2 is not set # CONFIG_MVIAC7 is not set # CONFIG_MCORE2 is not set # CONFIG_MATOM is not set # CONFIG_X86_GENERIC is not set CONFIG_X86_INTERNODE_CACHE_SHIFT=5 CONFIG_X86_L1_CACHE_SHIFT=5 CONFIG_X86_USE_PPRO_CHECKSUM=y CONFIG_X86_USE_3DNOW=y CONFIG_X86_TSC=y CONFIG_X86_CMPXCHG64=y CONFIG_X86_CMOV=y CONFIG_X86_MINIMUM_CPU_FAMILY=5 CONFIG_X86_DEBUGCTLMSR=y CONFIG_PROCESSOR_SELECT=y CONFIG_CPU_SUP_INTEL=y # CONFIG_CPU_SUP_CYRIX_32 is not set # CONFIG_CPU_SUP_AMD is not set CONFIG_CPU_SUP_CENTAUR=y # CONFIG_CPU_SUP_TRANSMETA_32 is not set CONFIG_CPU_SUP_UMC_32=y # CONFIG_HPET_TIMER is not set CONFIG_DMI=y CONFIG_NR_CPUS=1 # CONFIG_PREEMPT_NONE is not set # CONFIG_PREEMPT_VOLUNTARY is not set CONFIG_PREEMPT=y CONFIG_PREEMPT_COUNT=y CONFIG_X86_UP_APIC=y CONFIG_X86_UP_IOAPIC=y CONFIG_X86_LOCAL_APIC=y CONFIG_X86_IO_APIC=y # CONFIG_X86_REROUTE_FOR_BROKEN_BOOT_IRQS is not set CONFIG_X86_MCE=y # CONFIG_X86_MCE_INTEL is not set # CONFIG_X86_MCE_AMD is not set # CONFIG_X86_ANCIENT_MCE is not set # CONFIG_X86_MCE_INJECT is not set CONFIG_VM86=y CONFIG_TOSHIBA=m CONFIG_I8K=m # CONFIG_X86_REBOOTFIXUPS is not set CONFIG_MICROCODE=m CONFIG_MICROCODE_INTEL=y CONFIG_MICROCODE_AMD=y CONFIG_MICROCODE_OLD_INTERFACE=y CONFIG_MICROCODE_INTEL_LIB=y # CONFIG_X86_MSR is not set CONFIG_X86_CPUID=y CONFIG_NOHIGHMEM=y # CONFIG_HIGHMEM4G is not set # CONFIG_HIGHMEM64G is not set # CONFIG_VMSPLIT_3G is not set CONFIG_VMSPLIT_2G=y # CONFIG_VMSPLIT_1G is not set CONFIG_PAGE_OFFSET=0x80000000 CONFIG_X86_PAE=y CONFIG_ARCH_PHYS_ADDR_T_64BIT=y CONFIG_ARCH_FLATMEM_ENABLE=y CONFIG_ARCH_SPARSEMEM_ENABLE=y CONFIG_ARCH_SELECT_MEMORY_MODEL=y CONFIG_ILLEGAL_POINTER_VALUE=0 CONFIG_SELECT_MEMORY_MODEL=y CONFIG_FLATMEM_MANUAL=y # CONFIG_SPARSEMEM_MANUAL is not set CONFIG_FLATMEM=y CONFIG_FLAT_NODE_MEM_MAP=y CONFIG_SPARSEMEM_STATIC=y CONFIG_HAVE_MEMBLOCK=y CONFIG_HAVE_MEMBLOCK_NODE_MAP=y CONFIG_ARCH_DISCARD_MEMBLOCK=y # CONFIG_HAVE_BOOTMEM_INFO_NODE is not set CONFIG_PAGEFLAGS_EXTENDED=y CONFIG_SPLIT_PTLOCK_CPUS=999999 CONFIG_COMPACTION=y CONFIG_MIGRATION=y CONFIG_PHYS_ADDR_T_64BIT=y CONFIG_ZONE_DMA_FLAG=1 CONFIG_VIRT_TO_BUS=y # CONFIG_KSM is not set CONFIG_DEFAULT_MMAP_MIN_ADDR=4096 CONFIG_ARCH_SUPPORTS_MEMORY_FAILURE=y # CONFIG_MEMORY_FAILURE is not set CONFIG_TRANSPARENT_HUGEPAGE=y # CONFIG_TRANSPARENT_HUGEPAGE_ALWAYS is not set CONFIG_TRANSPARENT_HUGEPAGE_MADVISE=y # CONFIG_CROSS_MEMORY_ATTACH is not set CONFIG_NEED_PER_CPU_KM=y CONFIG_CLEANCACHE=y CONFIG_X86_CHECK_BIOS_CORRUPTION=y # CONFIG_X86_BOOTPARAM_MEMORY_CORRUPTION_CHECK is not set CONFIG_X86_RESERVE_LOW=64 CONFIG_MATH_EMULATION=y # CONFIG_MTRR is not set CONFIG_ARCH_RANDOM=y CONFIG_X86_SMAP=y CONFIG_SECCOMP=y # CONFIG_CC_STACKPROTECTOR is not set # CONFIG_HZ_100 is not set # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set CONFIG_HZ_1000=y CONFIG_HZ=1000 CONFIG_SCHED_HRTICK=y CONFIG_KEXEC=y CONFIG_PHYSICAL_START=0x1000000 # CONFIG_RELOCATABLE is not set CONFIG_PHYSICAL_ALIGN=0x1000000 CONFIG_COMPAT_VDSO=y CONFIG_CMDLINE_BOOL=y CONFIG_CMDLINE="" # CONFIG_CMDLINE_OVERRIDE is not set # # Power management and ACPI options # # CONFIG_SUSPEND is not set CONFIG_PM_RUNTIME=y CONFIG_PM=y # CONFIG_PM_DEBUG is not set # CONFIG_ACPI is not set CONFIG_SFI=y # # CPU Frequency scaling # CONFIG_CPU_FREQ=y CONFIG_CPU_FREQ_TABLE=y CONFIG_CPU_FREQ_GOV_COMMON=y # CONFIG_CPU_FREQ_STAT is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_POWERSAVE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND=y # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set CONFIG_CPU_FREQ_GOV_PERFORMANCE=y # CONFIG_CPU_FREQ_GOV_POWERSAVE is not set CONFIG_CPU_FREQ_GOV_USERSPACE=y CONFIG_CPU_FREQ_GOV_ONDEMAND=y # CONFIG_CPU_FREQ_GOV_CONSERVATIVE is not set # # x86 CPU frequency scaling drivers # CONFIG_X86_INTEL_PSTATE=y CONFIG_X86_POWERNOW_K6=y # CONFIG_X86_POWERNOW_K7 is not set CONFIG_X86_GX_SUSPMOD=y # CONFIG_X86_SPEEDSTEP_CENTRINO is not set CONFIG_X86_SPEEDSTEP_ICH=y # CONFIG_X86_SPEEDSTEP_SMI is not set CONFIG_X86_P4_CLOCKMOD=m CONFIG_X86_CPUFREQ_NFORCE2=y CONFIG_X86_LONGRUN=y # CONFIG_X86_E_POWERSAVER is not set # # shared options # CONFIG_X86_SPEEDSTEP_LIB=y CONFIG_X86_SPEEDSTEP_RELAXED_CAP_CHECK=y CONFIG_CPU_IDLE=y CONFIG_CPU_IDLE_MULTIPLE_DRIVERS=y CONFIG_CPU_IDLE_GOV_LADDER=y # CONFIG_ARCH_NEEDS_CPU_IDLE_COUPLED is not set # CONFIG_INTEL_IDLE is not set # # Bus options (PCI etc.) # CONFIG_PCI=y CONFIG_PCI_GOBIOS=y # CONFIG_PCI_GOMMCONFIG is not set # CONFIG_PCI_GODIRECT is not set # CONFIG_PCI_GOANY is not set CONFIG_PCI_BIOS=y CONFIG_PCI_DOMAINS=y # CONFIG_PCI_CNB20LE_QUIRK is not set CONFIG_PCIEPORTBUS=y CONFIG_HOTPLUG_PCI_PCIE=m # CONFIG_PCIEAER is not set CONFIG_PCIEASPM=y # CONFIG_PCIEASPM_DEBUG is not set CONFIG_PCIEASPM_DEFAULT=y # CONFIG_PCIEASPM_POWERSAVE is not set # CONFIG_PCIEASPM_PERFORMANCE is not set CONFIG_ARCH_SUPPORTS_MSI=y # CONFIG_PCI_MSI is not set # CONFIG_PCI_DEBUG is not set CONFIG_PCI_REALLOC_ENABLE_AUTO=y # CONFIG_PCI_STUB is not set # CONFIG_HT_IRQ is not set CONFIG_PCI_ATS=y # CONFIG_PCI_IOV is not set # CONFIG_PCI_PRI is not set CONFIG_PCI_PASID=y CONFIG_PCI_LABEL=y CONFIG_ISA_DMA_API=y # CONFIG_ISA is not set # CONFIG_SCx200 is not set CONFIG_ALIX=y # CONFIG_NET5501 is not set # CONFIG_GEOS is not set CONFIG_PCCARD=m CONFIG_PCMCIA=m CONFIG_PCMCIA_LOAD_CIS=y # CONFIG_CARDBUS is not set # # PC-card bridges # # CONFIG_YENTA is not set CONFIG_PD6729=m CONFIG_I82092=m CONFIG_PCCARD_NONSTATIC=y CONFIG_HOTPLUG_PCI=y CONFIG_HOTPLUG_PCI_COMPAQ=y CONFIG_HOTPLUG_PCI_COMPAQ_NVRAM=y # CONFIG_HOTPLUG_PCI_IBM is not set # CONFIG_HOTPLUG_PCI_CPCI is not set CONFIG_HOTPLUG_PCI_SHPC=y # CONFIG_RAPIDIO is not set # # Executable file formats / Emulations # CONFIG_BINFMT_ELF=y CONFIG_ARCH_BINFMT_ELF_RANDOMIZE_PIE=y CONFIG_CORE_DUMP_DEFAULT_ELF_HEADERS=y CONFIG_HAVE_AOUT=y CONFIG_BINFMT_AOUT=y CONFIG_BINFMT_MISC=y CONFIG_COREDUMP=y CONFIG_HAVE_ATOMIC_IOMAP=y CONFIG_HAVE_TEXT_POKE_SMP=y # CONFIG_NET is not set # # Device Drivers # # # Generic Driver Options # CONFIG_UEVENT_HELPER_PATH="" CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y CONFIG_STANDALONE=y CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=y CONFIG_FIRMWARE_IN_KERNEL=y CONFIG_EXTRA_FIRMWARE="" CONFIG_FW_LOADER_USER_HELPER=y # CONFIG_DEBUG_DRIVER is not set CONFIG_DEBUG_DEVRES=y # CONFIG_SYS_HYPERVISOR is not set # CONFIG_GENERIC_CPU_DEVICES is not set CONFIG_REGMAP=y CONFIG_REGMAP_I2C=y CONFIG_REGMAP_SPI=y CONFIG_REGMAP_MMIO=y CONFIG_REGMAP_IRQ=y CONFIG_DMA_SHARED_BUFFER=y # CONFIG_CMA is not set # # Bus devices # # CONFIG_MTD is not set CONFIG_PARPORT=y CONFIG_PARPORT_PC=m # CONFIG_PARPORT_SERIAL is not set CONFIG_PARPORT_PC_FIFO=y CONFIG_PARPORT_PC_SUPERIO=y # CONFIG_PARPORT_PC_PCMCIA is not set # CONFIG_PARPORT_GSC is not set CONFIG_PARPORT_AX88796=m CONFIG_PARPORT_1284=y CONFIG_PARPORT_NOT_PC=y # # Misc devices # CONFIG_SENSORS_LIS3LV02D=m # CONFIG_AD525X_DPOT is not set CONFIG_IBM_ASM=m CONFIG_PHANTOM=y # CONFIG_INTEL_MID_PTI is not set CONFIG_SGI_IOC4=y CONFIG_TIFM_CORE=y CONFIG_TIFM_7XX1=y # CONFIG_ICS932S401 is not set CONFIG_ATMEL_SSC=y CONFIG_ENCLOSURE_SERVICES=m # CONFIG_HP_ILO is not set # CONFIG_APDS9802ALS is not set CONFIG_ISL29003=m CONFIG_ISL29020=m CONFIG_SENSORS_TSL2550=m CONFIG_SENSORS_BH1780=m CONFIG_SENSORS_BH1770=m # CONFIG_SENSORS_APDS990X is not set CONFIG_HMC6352=m CONFIG_DS1682=y # CONFIG_TI_DAC7512 is not set # CONFIG_VMWARE_BALLOON is not set CONFIG_BMP085=y CONFIG_BMP085_I2C=y CONFIG_BMP085_SPI=m CONFIG_PCH_PHUB=m CONFIG_USB_SWITCH_FSA9480=y CONFIG_LATTICE_ECP3_CONFIG=m CONFIG_C2PORT=m CONFIG_C2PORT_DURAMAR_2150=m # # EEPROM support # CONFIG_EEPROM_AT24=m CONFIG_EEPROM_AT25=m CONFIG_EEPROM_LEGACY=m CONFIG_EEPROM_MAX6875=m # CONFIG_EEPROM_93CX6 is not set # CONFIG_EEPROM_93XX46 is not set CONFIG_CB710_CORE=y # CONFIG_CB710_DEBUG is not set CONFIG_CB710_DEBUG_ASSUMPTIONS=y # # Texas Instruments shared transport line discipline # CONFIG_SENSORS_LIS3_SPI=m CONFIG_SENSORS_LIS3_I2C=m # # Altera FPGA firmware download module # CONFIG_ALTERA_STAPL=m # CONFIG_VMWARE_VMCI is not set CONFIG_HAVE_IDE=y # # SCSI device support # CONFIG_SCSI_MOD=y # CONFIG_SCSI_DMA is not set # CONFIG_SCSI_NETLINK is not set # CONFIG_FUSION is not set # # IEEE 1394 (FireWire) support # # CONFIG_FIREWIRE is not set CONFIG_FIREWIRE_NOSY=y CONFIG_I2O=m CONFIG_I2O_LCT_NOTIFY_ON_CHANGES=y # CONFIG_I2O_EXT_ADAPTEC is not set # CONFIG_I2O_CONFIG is not set # CONFIG_I2O_BUS is not set CONFIG_I2O_PROC=m # CONFIG_MACINTOSH_DRIVERS is not set # # Input device support # CONFIG_INPUT=m CONFIG_INPUT_FF_MEMLESS=m CONFIG_INPUT_POLLDEV=m CONFIG_INPUT_SPARSEKMAP=m CONFIG_INPUT_MATRIXKMAP=m # # Userland interfaces # # CONFIG_INPUT_MOUSEDEV is not set CONFIG_INPUT_JOYDEV=m CONFIG_INPUT_EVDEV=m # CONFIG_INPUT_EVBUG is not set # # Input Device Drivers # # CONFIG_INPUT_KEYBOARD is not set # CONFIG_INPUT_MOUSE is not set CONFIG_INPUT_JOYSTICK=y CONFIG_JOYSTICK_ANALOG=m # CONFIG_JOYSTICK_A3D is not set CONFIG_JOYSTICK_ADI=m CONFIG_JOYSTICK_COBRA=m CONFIG_JOYSTICK_GF2K=m CONFIG_JOYSTICK_GRIP=m CONFIG_JOYSTICK_GRIP_MP=m # CONFIG_JOYSTICK_GUILLEMOT is not set # CONFIG_JOYSTICK_INTERACT is not set CONFIG_JOYSTICK_SIDEWINDER=m CONFIG_JOYSTICK_TMDC=m # CONFIG_JOYSTICK_IFORCE is not set CONFIG_JOYSTICK_WARRIOR=m # CONFIG_JOYSTICK_MAGELLAN is not set CONFIG_JOYSTICK_SPACEORB=m # CONFIG_JOYSTICK_SPACEBALL is not set # CONFIG_JOYSTICK_STINGER is not set # CONFIG_JOYSTICK_TWIDJOY is not set CONFIG_JOYSTICK_ZHENHUA=m # CONFIG_JOYSTICK_DB9 is not set CONFIG_JOYSTICK_GAMECON=m CONFIG_JOYSTICK_TURBOGRAFX=m CONFIG_JOYSTICK_AS5011=m CONFIG_JOYSTICK_JOYDUMP=m CONFIG_JOYSTICK_XPAD=m CONFIG_JOYSTICK_XPAD_FF=y CONFIG_JOYSTICK_XPAD_LEDS=y CONFIG_JOYSTICK_WALKERA0701=m CONFIG_INPUT_TABLET=y # CONFIG_TABLET_USB_ACECAD is not set CONFIG_TABLET_USB_AIPTEK=m CONFIG_TABLET_USB_GTCO=m CONFIG_TABLET_USB_HANWANG=m # CONFIG_TABLET_USB_KBTAB is not set CONFIG_TABLET_USB_WACOM=m CONFIG_INPUT_TOUCHSCREEN=y # CONFIG_TOUCHSCREEN_ADS7846 is not set CONFIG_TOUCHSCREEN_AD7877=m # CONFIG_TOUCHSCREEN_AD7879 is not set CONFIG_TOUCHSCREEN_ATMEL_MXT=m # CONFIG_TOUCHSCREEN_AUO_PIXCIR is not set # CONFIG_TOUCHSCREEN_BU21013 is not set CONFIG_TOUCHSCREEN_CY8CTMG110=m CONFIG_TOUCHSCREEN_CYTTSP_CORE=m CONFIG_TOUCHSCREEN_CYTTSP_I2C=m CONFIG_TOUCHSCREEN_CYTTSP_SPI=m # CONFIG_TOUCHSCREEN_DYNAPRO is not set # CONFIG_TOUCHSCREEN_HAMPSHIRE is not set # CONFIG_TOUCHSCREEN_EETI is not set CONFIG_TOUCHSCREEN_FUJITSU=m # CONFIG_TOUCHSCREEN_ILI210X is not set CONFIG_TOUCHSCREEN_GUNZE=m # CONFIG_TOUCHSCREEN_ELO is not set CONFIG_TOUCHSCREEN_WACOM_W8001=m # CONFIG_TOUCHSCREEN_WACOM_I2C is not set # CONFIG_TOUCHSCREEN_MAX11801 is not set CONFIG_TOUCHSCREEN_MCS5000=m # CONFIG_TOUCHSCREEN_MMS114 is not set # CONFIG_TOUCHSCREEN_MTOUCH is not set CONFIG_TOUCHSCREEN_INEXIO=m CONFIG_TOUCHSCREEN_MK712=m CONFIG_TOUCHSCREEN_PENMOUNT=m # CONFIG_TOUCHSCREEN_EDT_FT5X06 is not set # CONFIG_TOUCHSCREEN_TOUCHRIGHT is not set CONFIG_TOUCHSCREEN_TOUCHWIN=m # CONFIG_TOUCHSCREEN_TI_AM335X_TSC is not set CONFIG_TOUCHSCREEN_PIXCIR=m CONFIG_TOUCHSCREEN_WM831X=m CONFIG_TOUCHSCREEN_USB_COMPOSITE=m CONFIG_TOUCHSCREEN_MC13783=m CONFIG_TOUCHSCREEN_USB_EGALAX=y CONFIG_TOUCHSCREEN_USB_PANJIT=y CONFIG_TOUCHSCREEN_USB_3M=y CONFIG_TOUCHSCREEN_USB_ITM=y CONFIG_TOUCHSCREEN_USB_ETURBO=y CONFIG_TOUCHSCREEN_USB_GUNZE=y CONFIG_TOUCHSCREEN_USB_DMC_TSC10=y CONFIG_TOUCHSCREEN_USB_IRTOUCH=y # CONFIG_TOUCHSCREEN_USB_IDEALTEK is not set CONFIG_TOUCHSCREEN_USB_GENERAL_TOUCH=y # CONFIG_TOUCHSCREEN_USB_GOTOP is not set # CONFIG_TOUCHSCREEN_USB_JASTEC is not set CONFIG_TOUCHSCREEN_USB_ELO=y CONFIG_TOUCHSCREEN_USB_E2I=y # CONFIG_TOUCHSCREEN_USB_ZYTRONIC is not set CONFIG_TOUCHSCREEN_USB_ETT_TC45USB=y # CONFIG_TOUCHSCREEN_USB_NEXIO is not set CONFIG_TOUCHSCREEN_USB_EASYTOUCH=y # CONFIG_TOUCHSCREEN_TOUCHIT213 is not set CONFIG_TOUCHSCREEN_TSC_SERIO=m CONFIG_TOUCHSCREEN_TSC2005=m CONFIG_TOUCHSCREEN_TSC2007=m CONFIG_TOUCHSCREEN_PCAP=m CONFIG_TOUCHSCREEN_ST1232=m # CONFIG_TOUCHSCREEN_TPS6507X is not set CONFIG_INPUT_MISC=y # CONFIG_INPUT_88PM80X_ONKEY is not set CONFIG_INPUT_AD714X=m CONFIG_INPUT_AD714X_I2C=m # CONFIG_INPUT_AD714X_SPI is not set CONFIG_INPUT_ARIZONA_HAPTICS=m CONFIG_INPUT_BMA150=m CONFIG_INPUT_MC13783_PWRBUTTON=m CONFIG_INPUT_MMA8450=m # CONFIG_INPUT_MPU3050 is not set CONFIG_INPUT_APANEL=m CONFIG_INPUT_GP2A=m # CONFIG_INPUT_GPIO_TILT_POLLED is not set CONFIG_INPUT_WISTRON_BTNS=m CONFIG_INPUT_ATI_REMOTE2=m CONFIG_INPUT_KEYSPAN_REMOTE=m # CONFIG_INPUT_KXTJ9 is not set CONFIG_INPUT_POWERMATE=m CONFIG_INPUT_YEALINK=m # CONFIG_INPUT_CM109 is not set # CONFIG_INPUT_RETU_PWRBUTTON is not set # CONFIG_INPUT_TWL4030_PWRBUTTON is not set CONFIG_INPUT_TWL4030_VIBRA=m CONFIG_INPUT_UINPUT=m # CONFIG_INPUT_PCF50633_PMU is not set CONFIG_INPUT_PCF8574=m CONFIG_INPUT_PWM_BEEPER=m CONFIG_INPUT_GPIO_ROTARY_ENCODER=m # CONFIG_INPUT_WM831X_ON is not set CONFIG_INPUT_PCAP=m CONFIG_INPUT_ADXL34X=m CONFIG_INPUT_ADXL34X_I2C=m # CONFIG_INPUT_ADXL34X_SPI is not set CONFIG_INPUT_CMA3000=m CONFIG_INPUT_CMA3000_I2C=m # # Hardware I/O ports # CONFIG_SERIO=m CONFIG_SERIO_I8042=m CONFIG_SERIO_SERPORT=m # CONFIG_SERIO_CT82C710 is not set CONFIG_SERIO_PARKBD=m CONFIG_SERIO_PCIPS2=m CONFIG_SERIO_LIBPS2=m CONFIG_SERIO_RAW=m # CONFIG_SERIO_ALTERA_PS2 is not set CONFIG_SERIO_PS2MULT=m # CONFIG_SERIO_ARC_PS2 is not set CONFIG_GAMEPORT=y CONFIG_GAMEPORT_NS558=y CONFIG_GAMEPORT_L4=m # CONFIG_GAMEPORT_EMU10K1 is not set CONFIG_GAMEPORT_FM801=y # # Character devices # CONFIG_TTY=y # CONFIG_VT is not set CONFIG_UNIX98_PTYS=y CONFIG_DEVPTS_MULTIPLE_INSTANCES=y # CONFIG_LEGACY_PTYS is not set CONFIG_SERIAL_NONSTANDARD=y CONFIG_ROCKETPORT=m # CONFIG_CYCLADES is not set CONFIG_MOXA_INTELLIO=m # CONFIG_MOXA_SMARTIO is not set # CONFIG_SYNCLINK is not set # CONFIG_SYNCLINKMP is not set CONFIG_SYNCLINK_GT=y CONFIG_NOZOMI=m CONFIG_ISI=m CONFIG_N_HDLC=y CONFIG_TRACE_ROUTER=m CONFIG_TRACE_SINK=m CONFIG_DEVKMEM=y # CONFIG_STALDRV is not set # # Serial drivers # CONFIG_SERIAL_8250=m CONFIG_FIX_EARLYCON_MEM=y CONFIG_SERIAL_8250_PCI=m CONFIG_SERIAL_8250_CS=m CONFIG_SERIAL_8250_NR_UARTS=4 CONFIG_SERIAL_8250_RUNTIME_UARTS=4 # CONFIG_SERIAL_8250_EXTENDED is not set CONFIG_SERIAL_8250_DW=m # # Non-8250 serial port support # CONFIG_SERIAL_KGDB_NMI=y CONFIG_SERIAL_MAX3100=m CONFIG_SERIAL_MAX310X=y # CONFIG_SERIAL_MFD_HSU is not set # CONFIG_SERIAL_UARTLITE is not set CONFIG_SERIAL_CORE=y CONFIG_CONSOLE_POLL=y # CONFIG_SERIAL_JSM is not set # CONFIG_SERIAL_SCCNXP is not set CONFIG_SERIAL_TIMBERDALE=m CONFIG_SERIAL_ALTERA_JTAGUART=y # CONFIG_SERIAL_ALTERA_JTAGUART_CONSOLE is not set CONFIG_SERIAL_ALTERA_UART=m CONFIG_SERIAL_ALTERA_UART_MAXPORTS=4 CONFIG_SERIAL_ALTERA_UART_BAUDRATE=115200 # CONFIG_SERIAL_IFX6X60 is not set CONFIG_SERIAL_PCH_UART=y # CONFIG_SERIAL_PCH_UART_CONSOLE is not set CONFIG_SERIAL_ARC=y # CONFIG_SERIAL_ARC_CONSOLE is not set CONFIG_SERIAL_ARC_NR_PORTS=1 CONFIG_SERIAL_RP2=y CONFIG_SERIAL_RP2_NR_UARTS=32 # CONFIG_TTY_PRINTK is not set CONFIG_PRINTER=m CONFIG_LP_CONSOLE=y # CONFIG_PPDEV is not set CONFIG_HVC_DRIVER=y CONFIG_VIRTIO_CONSOLE=m CONFIG_IPMI_HANDLER=y CONFIG_IPMI_PANIC_EVENT=y # CONFIG_IPMI_PANIC_STRING is not set CONFIG_IPMI_DEVICE_INTERFACE=y CONFIG_IPMI_SI=m CONFIG_IPMI_WATCHDOG=y CONFIG_IPMI_POWEROFF=m CONFIG_HW_RANDOM=m CONFIG_HW_RANDOM_TIMERIOMEM=m CONFIG_HW_RANDOM_INTEL=m CONFIG_HW_RANDOM_AMD=m # CONFIG_HW_RANDOM_GEODE is not set CONFIG_HW_RANDOM_VIA=m CONFIG_HW_RANDOM_VIRTIO=m CONFIG_HW_RANDOM_TPM=m # CONFIG_NVRAM is not set CONFIG_R3964=y CONFIG_APPLICOM=m CONFIG_SONYPI=m # # PCMCIA character devices # # CONFIG_SYNCLINK_CS is not set CONFIG_CARDMAN_4000=m # CONFIG_CARDMAN_4040 is not set CONFIG_MWAVE=m CONFIG_PC8736x_GPIO=y CONFIG_NSC_GPIO=y # CONFIG_HANGCHECK_TIMER is not set CONFIG_TCG_TPM=y CONFIG_TCG_TIS=y # CONFIG_TCG_TIS_I2C_INFINEON is not set CONFIG_TCG_NSC=m CONFIG_TCG_ATMEL=y # CONFIG_TCG_ST33_I2C is not set # CONFIG_TELCLOCK is not set CONFIG_DEVPORT=y CONFIG_I2C=y CONFIG_I2C_BOARDINFO=y CONFIG_I2C_COMPAT=y # CONFIG_I2C_CHARDEV is not set CONFIG_I2C_MUX=m # # Multiplexer I2C Chip support # # CONFIG_I2C_MUX_GPIO is not set CONFIG_I2C_MUX_PCA9541=m CONFIG_I2C_MUX_PCA954x=m # CONFIG_I2C_HELPER_AUTO is not set CONFIG_I2C_SMBUS=m # # I2C Algorithms # CONFIG_I2C_ALGOBIT=y # CONFIG_I2C_ALGOPCF is not set CONFIG_I2C_ALGOPCA=m # # I2C Hardware Bus support # # # PC SMBus host controller drivers # CONFIG_I2C_ALI1535=m CONFIG_I2C_ALI1563=y CONFIG_I2C_ALI15X3=m CONFIG_I2C_AMD756=y CONFIG_I2C_AMD756_S4882=y # CONFIG_I2C_AMD8111 is not set # CONFIG_I2C_I801 is not set # CONFIG_I2C_ISCH is not set # CONFIG_I2C_PIIX4 is not set CONFIG_I2C_NFORCE2=y # CONFIG_I2C_NFORCE2_S4985 is not set # CONFIG_I2C_SIS5595 is not set CONFIG_I2C_SIS630=y CONFIG_I2C_SIS96X=y CONFIG_I2C_VIA=m # CONFIG_I2C_VIAPRO is not set # # I2C system bus drivers (mostly embedded / system-on-chip) # # CONFIG_I2C_CBUS_GPIO is not set # CONFIG_I2C_DESIGNWARE_PCI is not set # CONFIG_I2C_EG20T is not set # CONFIG_I2C_GPIO is not set # CONFIG_I2C_INTEL_MID is not set CONFIG_I2C_OCORES=m CONFIG_I2C_PCA_PLATFORM=m # CONFIG_I2C_PXA_PCI is not set CONFIG_I2C_SIMTEC=y CONFIG_I2C_XILINX=m # # External I2C/SMBus adapter drivers # # CONFIG_I2C_DIOLAN_U2C is not set # CONFIG_I2C_PARPORT is not set CONFIG_I2C_PARPORT_LIGHT=m CONFIG_I2C_TAOS_EVM=m CONFIG_I2C_TINY_USB=m # # Other I2C/SMBus bus drivers # # CONFIG_SCx200_ACB is not set CONFIG_I2C_STUB=m # CONFIG_I2C_DEBUG_CORE is not set # CONFIG_I2C_DEBUG_ALGO is not set # CONFIG_I2C_DEBUG_BUS is not set CONFIG_SPI=y # CONFIG_SPI_DEBUG is not set CONFIG_SPI_MASTER=y # # SPI Master Controller Drivers # CONFIG_SPI_ALTERA=m CONFIG_SPI_BITBANG=y # CONFIG_SPI_BUTTERFLY is not set CONFIG_SPI_GPIO=y # CONFIG_SPI_LM70_LLP is not set CONFIG_SPI_OC_TINY=m CONFIG_SPI_PXA2XX_DMA=y CONFIG_SPI_PXA2XX=m CONFIG_SPI_PXA2XX_PCI=m CONFIG_SPI_SC18IS602=y CONFIG_SPI_TOPCLIFF_PCH=y CONFIG_SPI_XCOMM=m CONFIG_SPI_XILINX=m CONFIG_SPI_DESIGNWARE=m # CONFIG_SPI_DW_PCI is not set # # SPI Protocol Masters # CONFIG_SPI_SPIDEV=m CONFIG_SPI_TLE62X0=y CONFIG_HSI=m CONFIG_HSI_BOARDINFO=y # # HSI clients # CONFIG_HSI_CHAR=m # # PPS support # CONFIG_PPS=m # CONFIG_PPS_DEBUG is not set CONFIG_NTP_PPS=y # # PPS clients support # CONFIG_PPS_CLIENT_KTIMER=m CONFIG_PPS_CLIENT_LDISC=m CONFIG_PPS_CLIENT_PARPORT=m CONFIG_PPS_CLIENT_GPIO=m # # PPS generators support # # # PTP clock support # CONFIG_PTP_1588_CLOCK=m # # Enable PHYLIB and NETWORK_PHY_TIMESTAMPING to see the additional clocks. # CONFIG_PTP_1588_CLOCK_PCH=m CONFIG_ARCH_WANT_OPTIONAL_GPIOLIB=y CONFIG_GPIOLIB=y CONFIG_DEBUG_GPIO=y # CONFIG_GPIO_SYSFS is not set CONFIG_GPIO_GENERIC=y CONFIG_GPIO_MAX730X=y # # Memory mapped GPIO drivers: # CONFIG_GPIO_GENERIC_PLATFORM=y # CONFIG_GPIO_IT8761E is not set # CONFIG_GPIO_TS5500 is not set # CONFIG_GPIO_SCH is not set # CONFIG_GPIO_ICH is not set # CONFIG_GPIO_VX855 is not set # # I2C GPIO expanders: # CONFIG_GPIO_ARIZONA=y CONFIG_GPIO_MAX7300=y # CONFIG_GPIO_MAX732X is not set CONFIG_GPIO_PCA953X=m CONFIG_GPIO_PCF857X=y # CONFIG_GPIO_RC5T583 is not set CONFIG_GPIO_SX150X=y # CONFIG_GPIO_TC3589X is not set # CONFIG_GPIO_TWL4030 is not set CONFIG_GPIO_WM831X=y CONFIG_GPIO_WM8350=m # CONFIG_GPIO_WM8994 is not set CONFIG_GPIO_ADP5588=m # # PCI GPIO expanders: # CONFIG_GPIO_BT8XX=m CONFIG_GPIO_AMD8111=y # CONFIG_GPIO_LANGWELL is not set CONFIG_GPIO_PCH=y CONFIG_GPIO_ML_IOH=m # CONFIG_GPIO_TIMBERDALE is not set CONFIG_GPIO_RDC321X=m # # SPI GPIO expanders: # # CONFIG_GPIO_MAX7301 is not set CONFIG_GPIO_MCP23S08=y # CONFIG_GPIO_MC33880 is not set CONFIG_GPIO_74X164=y # # AC97 GPIO expanders: # # # MODULbus GPIO expanders: # # CONFIG_GPIO_JANZ_TTL is not set # CONFIG_GPIO_PALMAS is not set CONFIG_GPIO_TPS65910=y # # USB GPIO expanders: # CONFIG_W1=y # # 1-wire Bus Masters # # CONFIG_W1_MASTER_MATROX is not set CONFIG_W1_MASTER_DS2490=m # CONFIG_W1_MASTER_DS2482 is not set # CONFIG_W1_MASTER_DS1WM is not set # CONFIG_W1_MASTER_GPIO is not set # # 1-wire Slaves # CONFIG_W1_SLAVE_THERM=y CONFIG_W1_SLAVE_SMEM=m # CONFIG_W1_SLAVE_DS2408 is not set # CONFIG_W1_SLAVE_DS2423 is not set CONFIG_W1_SLAVE_DS2431=m CONFIG_W1_SLAVE_DS2433=m CONFIG_W1_SLAVE_DS2433_CRC=y CONFIG_W1_SLAVE_DS2760=y CONFIG_W1_SLAVE_DS2780=m CONFIG_W1_SLAVE_DS2781=y CONFIG_W1_SLAVE_DS28E04=m # CONFIG_W1_SLAVE_BQ27000 is not set CONFIG_POWER_SUPPLY=y CONFIG_POWER_SUPPLY_DEBUG=y CONFIG_PDA_POWER=m CONFIG_GENERIC_ADC_BATTERY=y CONFIG_WM831X_BACKUP=m # CONFIG_WM831X_POWER is not set CONFIG_WM8350_POWER=y CONFIG_TEST_POWER=m CONFIG_BATTERY_DS2760=y CONFIG_BATTERY_DS2780=m CONFIG_BATTERY_DS2781=y CONFIG_BATTERY_DS2782=y # CONFIG_BATTERY_SBS is not set # CONFIG_BATTERY_BQ27x00 is not set # CONFIG_BATTERY_MAX17040 is not set CONFIG_BATTERY_MAX17042=y CONFIG_CHARGER_PCF50633=m # CONFIG_BATTERY_RX51 is not set CONFIG_CHARGER_ISP1704=y CONFIG_CHARGER_MAX8903=y CONFIG_CHARGER_TWL4030=m CONFIG_CHARGER_LP8727=y CONFIG_CHARGER_GPIO=y # CONFIG_CHARGER_MANAGER is not set # CONFIG_CHARGER_MAX8997 is not set CONFIG_CHARGER_MAX8998=m CONFIG_CHARGER_BQ2415X=m CONFIG_CHARGER_SMB347=y CONFIG_BATTERY_GOLDFISH=y # CONFIG_POWER_RESET is not set CONFIG_POWER_AVS=y CONFIG_HWMON=m CONFIG_HWMON_VID=m # CONFIG_HWMON_DEBUG_CHIP is not set # # Native drivers # CONFIG_SENSORS_ABITUGURU=m CONFIG_SENSORS_ABITUGURU3=m CONFIG_SENSORS_AD7314=m CONFIG_SENSORS_AD7414=m # CONFIG_SENSORS_AD7418 is not set CONFIG_SENSORS_ADCXX=m CONFIG_SENSORS_ADM1021=m CONFIG_SENSORS_ADM1025=m # CONFIG_SENSORS_ADM1026 is not set CONFIG_SENSORS_ADM1029=m # CONFIG_SENSORS_ADM1031 is not set # CONFIG_SENSORS_ADM9240 is not set CONFIG_SENSORS_ADT7410=m CONFIG_SENSORS_ADT7411=m CONFIG_SENSORS_ADT7462=m # CONFIG_SENSORS_ADT7470 is not set CONFIG_SENSORS_ADT7475=m CONFIG_SENSORS_ASC7621=m CONFIG_SENSORS_K8TEMP=m CONFIG_SENSORS_K10TEMP=m CONFIG_SENSORS_FAM15H_POWER=m CONFIG_SENSORS_ASB100=m # CONFIG_SENSORS_ATXP1 is not set # CONFIG_SENSORS_DS620 is not set # CONFIG_SENSORS_DS1621 is not set # CONFIG_SENSORS_I5K_AMB is not set CONFIG_SENSORS_F71805F=m CONFIG_SENSORS_F71882FG=m # CONFIG_SENSORS_F75375S is not set CONFIG_SENSORS_FSCHMD=m # CONFIG_SENSORS_G760A is not set # CONFIG_SENSORS_GL518SM is not set CONFIG_SENSORS_GL520SM=m # CONFIG_SENSORS_GPIO_FAN is not set CONFIG_SENSORS_HIH6130=m # CONFIG_SENSORS_CORETEMP is not set CONFIG_SENSORS_IBMAEM=m # CONFIG_SENSORS_IBMPEX is not set CONFIG_SENSORS_IT87=m CONFIG_SENSORS_JC42=m # CONFIG_SENSORS_LINEAGE is not set # CONFIG_SENSORS_LM63 is not set # CONFIG_SENSORS_LM70 is not set # CONFIG_SENSORS_LM73 is not set CONFIG_SENSORS_LM75=m # CONFIG_SENSORS_LM77 is not set # CONFIG_SENSORS_LM78 is not set CONFIG_SENSORS_LM80=m # CONFIG_SENSORS_LM83 is not set CONFIG_SENSORS_LM85=m CONFIG_SENSORS_LM87=m CONFIG_SENSORS_LM90=m CONFIG_SENSORS_LM92=m CONFIG_SENSORS_LM93=m # CONFIG_SENSORS_LTC4151 is not set # CONFIG_SENSORS_LTC4215 is not set CONFIG_SENSORS_LTC4245=m CONFIG_SENSORS_LTC4261=m # CONFIG_SENSORS_LM95241 is not set CONFIG_SENSORS_LM95245=m CONFIG_SENSORS_MAX1111=m CONFIG_SENSORS_MAX16065=m CONFIG_SENSORS_MAX1619=m # CONFIG_SENSORS_MAX1668 is not set CONFIG_SENSORS_MAX197=m # CONFIG_SENSORS_MAX6639 is not set CONFIG_SENSORS_MAX6642=m # CONFIG_SENSORS_MAX6650 is not set CONFIG_SENSORS_MAX6697=m CONFIG_SENSORS_MCP3021=m # CONFIG_SENSORS_NTC_THERMISTOR is not set CONFIG_SENSORS_PC87360=m CONFIG_SENSORS_PC87427=m # CONFIG_SENSORS_PCF8591 is not set # CONFIG_PMBUS is not set CONFIG_SENSORS_SHT15=m CONFIG_SENSORS_SHT21=m # CONFIG_SENSORS_SIS5595 is not set CONFIG_SENSORS_SMM665=m CONFIG_SENSORS_DME1737=m # CONFIG_SENSORS_EMC1403 is not set CONFIG_SENSORS_EMC2103=m # CONFIG_SENSORS_EMC6W201 is not set CONFIG_SENSORS_SMSC47M1=m # CONFIG_SENSORS_SMSC47M192 is not set CONFIG_SENSORS_SMSC47B397=m # CONFIG_SENSORS_SCH56XX_COMMON is not set CONFIG_SENSORS_ADS1015=m CONFIG_SENSORS_ADS7828=m # CONFIG_SENSORS_ADS7871 is not set # CONFIG_SENSORS_AMC6821 is not set # CONFIG_SENSORS_INA209 is not set CONFIG_SENSORS_INA2XX=m CONFIG_SENSORS_THMC50=m CONFIG_SENSORS_TMP102=m CONFIG_SENSORS_TMP401=m CONFIG_SENSORS_TMP421=m CONFIG_SENSORS_TWL4030_MADC=m CONFIG_SENSORS_VIA_CPUTEMP=m # CONFIG_SENSORS_VIA686A is not set CONFIG_SENSORS_VT1211=m CONFIG_SENSORS_VT8231=m CONFIG_SENSORS_W83781D=m # CONFIG_SENSORS_W83791D is not set # CONFIG_SENSORS_W83792D is not set CONFIG_SENSORS_W83793=m CONFIG_SENSORS_W83795=m CONFIG_SENSORS_W83795_FANCTRL=y # CONFIG_SENSORS_W83L785TS is not set CONFIG_SENSORS_W83L786NG=m # CONFIG_SENSORS_W83627HF is not set CONFIG_SENSORS_W83627EHF=m # CONFIG_SENSORS_WM831X is not set # CONFIG_SENSORS_WM8350 is not set # CONFIG_SENSORS_APPLESMC is not set CONFIG_SENSORS_MC13783_ADC=m CONFIG_THERMAL=y # CONFIG_THERMAL_DEFAULT_GOV_STEP_WISE is not set CONFIG_THERMAL_DEFAULT_GOV_FAIR_SHARE=y # CONFIG_THERMAL_DEFAULT_GOV_USER_SPACE is not set CONFIG_FAIR_SHARE=y # CONFIG_STEP_WISE is not set # CONFIG_USER_SPACE is not set # CONFIG_CPU_THERMAL is not set # CONFIG_WATCHDOG is not set CONFIG_SSB_POSSIBLE=y # # Sonics Silicon Backplane # CONFIG_SSB=y CONFIG_SSB_PCIHOST_POSSIBLE=y # CONFIG_SSB_PCIHOST is not set CONFIG_SSB_SILENT=y CONFIG_SSB_DRIVER_GPIO=y CONFIG_BCMA_POSSIBLE=y # # Broadcom specific AMBA # CONFIG_BCMA=m CONFIG_BCMA_HOST_PCI_POSSIBLE=y CONFIG_BCMA_HOST_PCI=y CONFIG_BCMA_DRIVER_GMAC_CMN=y CONFIG_BCMA_DRIVER_GPIO=y CONFIG_BCMA_DEBUG=y # # Multifunction device drivers # CONFIG_MFD_CORE=y # CONFIG_MFD_88PM860X is not set CONFIG_MFD_88PM800=y CONFIG_MFD_88PM805=m # CONFIG_MFD_SM501 is not set CONFIG_MFD_RTSX_PCI=m CONFIG_MFD_TI_AM335X_TSCADC=m # CONFIG_HTC_PASIC3 is not set CONFIG_HTC_I2CPLD=y # CONFIG_MFD_LM3533 is not set CONFIG_TPS6105X=m # CONFIG_TPS65010 is not set CONFIG_TPS6507X=m CONFIG_MFD_TPS65217=m # CONFIG_MFD_TPS6586X is not set CONFIG_MFD_TPS65910=y # CONFIG_MFD_TPS65912_I2C is not set # CONFIG_MFD_TPS65912_SPI is not set # CONFIG_MFD_TPS80031 is not set CONFIG_TWL4030_CORE=y CONFIG_TWL4030_MADC=m CONFIG_MFD_TWL4030_AUDIO=y # CONFIG_TWL6040_CORE is not set # CONFIG_MFD_STMPE is not set CONFIG_MFD_TC3589X=y # CONFIG_MFD_TMIO is not set CONFIG_MFD_SMSC=y # CONFIG_PMIC_DA903X is not set # CONFIG_MFD_DA9052_SPI is not set # CONFIG_MFD_DA9052_I2C is not set # CONFIG_MFD_DA9055 is not set # CONFIG_PMIC_ADP5520 is not set # CONFIG_MFD_LP8788 is not set CONFIG_MFD_MAX77686=y # CONFIG_MFD_MAX77693 is not set # CONFIG_MFD_MAX8907 is not set # CONFIG_MFD_MAX8925 is not set CONFIG_MFD_MAX8997=y CONFIG_MFD_MAX8998=y CONFIG_MFD_SEC_CORE=y CONFIG_MFD_ARIZONA=y # CONFIG_MFD_ARIZONA_I2C is not set CONFIG_MFD_ARIZONA_SPI=y CONFIG_MFD_WM5102=y # CONFIG_MFD_WM5110 is not set # CONFIG_MFD_WM8400 is not set CONFIG_MFD_WM831X=y CONFIG_MFD_WM831X_I2C=y # CONFIG_MFD_WM831X_SPI is not set CONFIG_MFD_WM8350=y CONFIG_MFD_WM8350_I2C=y CONFIG_MFD_WM8994=y CONFIG_MFD_PCF50633=m # CONFIG_PCF50633_ADC is not set CONFIG_PCF50633_GPIO=m CONFIG_MFD_MC13783=y CONFIG_MFD_MC13XXX=y CONFIG_MFD_MC13XXX_SPI=y CONFIG_MFD_MC13XXX_I2C=m # CONFIG_ABX500_CORE is not set CONFIG_EZX_PCAP=y # CONFIG_MFD_CS5535 is not set CONFIG_MFD_TIMBERDALE=y CONFIG_LPC_SCH=y CONFIG_LPC_ICH=y CONFIG_MFD_RDC321X=m CONFIG_MFD_JANZ_CMODIO=m CONFIG_MFD_VX855=y # CONFIG_MFD_WL1273_CORE is not set # CONFIG_MFD_TPS65090 is not set CONFIG_MFD_AAT2870_CORE=y CONFIG_MFD_RC5T583=y CONFIG_MFD_PALMAS=y # CONFIG_MFD_VIPERBOARD is not set CONFIG_MFD_RETU=m # CONFIG_MFD_AS3711 is not set CONFIG_REGULATOR=y # CONFIG_REGULATOR_DEBUG is not set # CONFIG_REGULATOR_DUMMY is not set CONFIG_REGULATOR_FIXED_VOLTAGE=m CONFIG_REGULATOR_VIRTUAL_CONSUMER=y # CONFIG_REGULATOR_USERSPACE_CONSUMER is not set # CONFIG_REGULATOR_GPIO is not set # CONFIG_REGULATOR_AD5398 is not set # CONFIG_REGULATOR_AAT2870 is not set CONFIG_REGULATOR_ARIZONA=y CONFIG_REGULATOR_FAN53555=m CONFIG_REGULATOR_MC13XXX_CORE=m CONFIG_REGULATOR_MC13783=m # CONFIG_REGULATOR_MC13892 is not set CONFIG_REGULATOR_ISL6271A=y CONFIG_REGULATOR_MAX1586=y # CONFIG_REGULATOR_MAX8649 is not set CONFIG_REGULATOR_MAX8660=m CONFIG_REGULATOR_MAX8952=m # CONFIG_REGULATOR_MAX8973 is not set CONFIG_REGULATOR_MAX8997=m CONFIG_REGULATOR_MAX8998=m CONFIG_REGULATOR_MAX77686=m # CONFIG_REGULATOR_PCAP is not set # CONFIG_REGULATOR_LP3971 is not set # CONFIG_REGULATOR_LP3972 is not set # CONFIG_REGULATOR_LP872X is not set CONFIG_REGULATOR_LP8755=m CONFIG_REGULATOR_PCF50633=m CONFIG_REGULATOR_RC5T583=m CONFIG_REGULATOR_S2MPS11=m CONFIG_REGULATOR_S5M8767=m CONFIG_REGULATOR_PALMAS=m # CONFIG_REGULATOR_TPS51632 is not set CONFIG_REGULATOR_TPS6105X=m # CONFIG_REGULATOR_TPS62360 is not set CONFIG_REGULATOR_TPS65023=y CONFIG_REGULATOR_TPS6507X=m # CONFIG_REGULATOR_TPS65217 is not set CONFIG_REGULATOR_TPS6524X=y CONFIG_REGULATOR_TPS65910=m CONFIG_REGULATOR_TWL4030=y CONFIG_REGULATOR_WM831X=y # CONFIG_REGULATOR_WM8350 is not set CONFIG_REGULATOR_WM8994=y CONFIG_MEDIA_SUPPORT=m # # Multimedia core support # CONFIG_MEDIA_CAMERA_SUPPORT=y # CONFIG_MEDIA_ANALOG_TV_SUPPORT is not set # CONFIG_MEDIA_DIGITAL_TV_SUPPORT is not set # CONFIG_MEDIA_RADIO_SUPPORT is not set # CONFIG_MEDIA_RC_SUPPORT is not set # CONFIG_MEDIA_CONTROLLER is not set CONFIG_VIDEO_DEV=m CONFIG_VIDEO_V4L2=m CONFIG_VIDEO_ADV_DEBUG=y # CONFIG_VIDEO_FIXED_MINOR_RANGES is not set CONFIG_VIDEO_TUNER=m CONFIG_V4L2_MEM2MEM_DEV=m CONFIG_VIDEOBUF_GEN=m CONFIG_VIDEOBUF_DMA_SG=m CONFIG_VIDEOBUF_VMALLOC=m CONFIG_VIDEOBUF2_CORE=m CONFIG_VIDEOBUF2_MEMOPS=m CONFIG_VIDEOBUF2_DMA_CONTIG=m CONFIG_VIDEOBUF2_VMALLOC=m # CONFIG_VIDEO_V4L2_INT_DEVICE is not set # CONFIG_TTPCI_EEPROM is not set # # Media drivers # CONFIG_MEDIA_USB_SUPPORT=y # # Webcam devices # # CONFIG_USB_VIDEO_CLASS is not set CONFIG_USB_GSPCA=m CONFIG_USB_M5602=m # CONFIG_USB_STV06XX is not set CONFIG_USB_GL860=m CONFIG_USB_GSPCA_BENQ=m CONFIG_USB_GSPCA_CONEX=m # CONFIG_USB_GSPCA_CPIA1 is not set # CONFIG_USB_GSPCA_ETOMS is not set CONFIG_USB_GSPCA_FINEPIX=m CONFIG_USB_GSPCA_JEILINJ=m CONFIG_USB_GSPCA_JL2005BCD=m CONFIG_USB_GSPCA_KINECT=m CONFIG_USB_GSPCA_KONICA=m CONFIG_USB_GSPCA_MARS=m # CONFIG_USB_GSPCA_MR97310A is not set CONFIG_USB_GSPCA_NW80X=m # CONFIG_USB_GSPCA_OV519 is not set CONFIG_USB_GSPCA_OV534=m CONFIG_USB_GSPCA_OV534_9=m CONFIG_USB_GSPCA_PAC207=m CONFIG_USB_GSPCA_PAC7302=m # CONFIG_USB_GSPCA_PAC7311 is not set # CONFIG_USB_GSPCA_SE401 is not set CONFIG_USB_GSPCA_SN9C2028=m CONFIG_USB_GSPCA_SN9C20X=m CONFIG_USB_GSPCA_SONIXB=m CONFIG_USB_GSPCA_SONIXJ=m CONFIG_USB_GSPCA_SPCA500=m CONFIG_USB_GSPCA_SPCA501=m # CONFIG_USB_GSPCA_SPCA505 is not set # CONFIG_USB_GSPCA_SPCA506 is not set CONFIG_USB_GSPCA_SPCA508=m CONFIG_USB_GSPCA_SPCA561=m # CONFIG_USB_GSPCA_SPCA1528 is not set # CONFIG_USB_GSPCA_SQ905 is not set CONFIG_USB_GSPCA_SQ905C=m # CONFIG_USB_GSPCA_SQ930X is not set # CONFIG_USB_GSPCA_STK014 is not set CONFIG_USB_GSPCA_STV0680=m # CONFIG_USB_GSPCA_SUNPLUS is not set CONFIG_USB_GSPCA_T613=m CONFIG_USB_GSPCA_TOPRO=m # CONFIG_USB_GSPCA_TV8532 is not set CONFIG_USB_GSPCA_VC032X=m # CONFIG_USB_GSPCA_VICAM is not set CONFIG_USB_GSPCA_XIRLINK_CIT=m CONFIG_USB_GSPCA_ZC3XX=m # CONFIG_USB_PWC is not set CONFIG_VIDEO_CPIA2=m CONFIG_USB_ZR364XX=m CONFIG_USB_STKWEBCAM=m CONFIG_USB_S2255=m # CONFIG_USB_SN9C102 is not set # # Webcam, TV (analog/digital) USB devices # CONFIG_VIDEO_EM28XX=m # CONFIG_VIDEO_EM28XX_ALSA is not set CONFIG_MEDIA_PCI_SUPPORT=y # # Media capture support # CONFIG_V4L_PLATFORM_DRIVERS=y # CONFIG_VIDEO_CAFE_CCIC is not set # CONFIG_SOC_CAMERA is not set # CONFIG_V4L_MEM2MEM_DRIVERS is not set CONFIG_V4L_TEST_DRIVERS=y CONFIG_VIDEO_MEM2MEM_TESTDEV=m # # Supported MMC/SDIO adapters # CONFIG_MEDIA_PARPORT_SUPPORT=y # CONFIG_VIDEO_BWQCAM is not set CONFIG_VIDEO_CQCAM=m CONFIG_VIDEO_W9966=m CONFIG_VIDEO_TVEEPROM=m # # Media ancillary drivers (tuners, sensors, i2c, frontends) # # CONFIG_MEDIA_SUBDRV_AUTOSELECT is not set # # Encoders, decoders, sensors and other helper chips # # # Audio decoders, processors and mixers # # CONFIG_VIDEO_TVAUDIO is not set CONFIG_VIDEO_TDA7432=m CONFIG_VIDEO_TDA9840=m # CONFIG_VIDEO_TEA6415C is not set CONFIG_VIDEO_TEA6420=m # CONFIG_VIDEO_MSP3400 is not set # CONFIG_VIDEO_CS5345 is not set CONFIG_VIDEO_CS53L32A=m CONFIG_VIDEO_TLV320AIC23B=m # CONFIG_VIDEO_WM8775 is not set CONFIG_VIDEO_WM8739=m CONFIG_VIDEO_VP27SMPX=m # # RDS decoders # CONFIG_VIDEO_SAA6588=m # # Video decoders # CONFIG_VIDEO_ADV7180=m CONFIG_VIDEO_ADV7183=m CONFIG_VIDEO_BT819=m CONFIG_VIDEO_BT856=m # CONFIG_VIDEO_BT866 is not set # CONFIG_VIDEO_KS0127 is not set # CONFIG_VIDEO_SAA7110 is not set # CONFIG_VIDEO_SAA711X is not set # CONFIG_VIDEO_SAA7191 is not set CONFIG_VIDEO_TVP514X=m CONFIG_VIDEO_TVP5150=m CONFIG_VIDEO_TVP7002=m CONFIG_VIDEO_VPX3220=m # # Video and audio decoders # # CONFIG_VIDEO_SAA717X is not set CONFIG_VIDEO_CX25840=m # # Video encoders # # CONFIG_VIDEO_SAA7127 is not set CONFIG_VIDEO_SAA7185=m CONFIG_VIDEO_ADV7170=m CONFIG_VIDEO_ADV7175=m CONFIG_VIDEO_ADV7343=m CONFIG_VIDEO_ADV7393=m # CONFIG_VIDEO_AK881X is not set # # Camera sensor devices # CONFIG_VIDEO_OV7670=m CONFIG_VIDEO_VS6624=m CONFIG_VIDEO_MT9V011=m CONFIG_VIDEO_SR030PC30=m # # Flash devices # # # Video improvement chips # # CONFIG_VIDEO_UPD64031A is not set CONFIG_VIDEO_UPD64083=m # # Miscelaneous helper chips # CONFIG_VIDEO_THS7303=m # CONFIG_VIDEO_M52790 is not set # # Sensors used on soc_camera driver # # # Customise DVB Frontends # # CONFIG_DVB_AU8522_V4L is not set CONFIG_DVB_TUNER_DIB0070=m CONFIG_DVB_TUNER_DIB0090=m # # Tools to develop new frontends # CONFIG_DVB_DUMMY_FE=m # # Graphics support # CONFIG_AGP=m CONFIG_AGP_ALI=m CONFIG_AGP_ATI=m # CONFIG_AGP_AMD is not set CONFIG_AGP_INTEL=m CONFIG_AGP_NVIDIA=m CONFIG_AGP_SIS=m CONFIG_AGP_SWORKS=m # CONFIG_AGP_VIA is not set # CONFIG_AGP_EFFICEON is not set # CONFIG_VGA_ARB is not set # CONFIG_DRM is not set # CONFIG_VGASTATE is not set CONFIG_VIDEO_OUTPUT_CONTROL=y # CONFIG_FB is not set CONFIG_EXYNOS_VIDEO=y CONFIG_BACKLIGHT_LCD_SUPPORT=y CONFIG_LCD_CLASS_DEVICE=y # CONFIG_LCD_L4F00242T03 is not set CONFIG_LCD_LMS283GF05=m CONFIG_LCD_LTV350QV=m CONFIG_LCD_ILI9320=y CONFIG_LCD_TDO24M=y CONFIG_LCD_VGG2432A4=y CONFIG_LCD_PLATFORM=m CONFIG_LCD_S6E63M0=m CONFIG_LCD_LD9040=m CONFIG_LCD_AMS369FG06=m # CONFIG_LCD_LMS501KF03 is not set # CONFIG_LCD_HX8357 is not set CONFIG_BACKLIGHT_CLASS_DEVICE=m CONFIG_BACKLIGHT_GENERIC=m CONFIG_BACKLIGHT_PWM=m CONFIG_BACKLIGHT_SAHARA=m # CONFIG_BACKLIGHT_WM831X is not set CONFIG_BACKLIGHT_ADP8860=m CONFIG_BACKLIGHT_ADP8870=m CONFIG_BACKLIGHT_PCF50633=m CONFIG_BACKLIGHT_AAT2870=m # CONFIG_BACKLIGHT_LM3630 is not set CONFIG_BACKLIGHT_LM3639=m # CONFIG_BACKLIGHT_LP855X is not set CONFIG_BACKLIGHT_PANDORA=m # CONFIG_BACKLIGHT_TPS65217 is not set CONFIG_SOUND=y CONFIG_SOUND_OSS_CORE=y # CONFIG_SOUND_OSS_CORE_PRECLAIM is not set CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_HWDEP=y CONFIG_SND_RAWMIDI=y CONFIG_SND_COMPRESS_OFFLOAD=y # CONFIG_SND_SEQUENCER is not set CONFIG_SND_OSSEMUL=y CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_PCM_OSS_PLUGINS=y CONFIG_SND_HRTIMER=m CONFIG_SND_DYNAMIC_MINORS=y CONFIG_SND_SUPPORT_OLD_API=y CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_DEBUG is not set CONFIG_SND_VMASTER=y CONFIG_SND_DMA_SGBUF=y # CONFIG_SND_RAWMIDI_SEQ is not set # CONFIG_SND_OPL3_LIB_SEQ is not set # CONFIG_SND_OPL4_LIB_SEQ is not set # CONFIG_SND_SBAWE_SEQ is not set # CONFIG_SND_EMU10K1_SEQ is not set CONFIG_SND_VX_LIB=m CONFIG_SND_DRIVERS=y # CONFIG_SND_DUMMY is not set CONFIG_SND_ALOOP=y # CONFIG_SND_MTPAV is not set CONFIG_SND_MTS64=y CONFIG_SND_SERIAL_U16550=m # CONFIG_SND_MPU401 is not set # CONFIG_SND_PORTMAN2X4 is not set # CONFIG_SND_PCI is not set # CONFIG_SND_SPI is not set CONFIG_SND_USB=y # CONFIG_SND_USB_AUDIO is not set CONFIG_SND_USB_UA101=m CONFIG_SND_USB_USX2Y=m CONFIG_SND_USB_CAIAQ=y # CONFIG_SND_USB_US122L is not set CONFIG_SND_USB_6FIRE=y CONFIG_SND_PCMCIA=y CONFIG_SND_VXPOCKET=m # CONFIG_SND_PDAUDIOCF is not set CONFIG_SND_SOC=y CONFIG_SND_ATMEL_SOC=y CONFIG_SND_SOC_I2C_AND_SPI=y CONFIG_SND_SOC_ALL_CODECS=y CONFIG_SND_SOC_ARIZONA=y CONFIG_SND_SOC_WM_HUBS=y CONFIG_SND_SOC_WM_ADSP=y CONFIG_SND_SOC_AD1836=y CONFIG_SND_SOC_AD193X=y CONFIG_SND_SOC_AD73311=y CONFIG_SND_SOC_ADAU1373=y CONFIG_SND_SOC_ADAV80X=y CONFIG_SND_SOC_ADS117X=y CONFIG_SND_SOC_AK4104=y CONFIG_SND_SOC_AK4535=y CONFIG_SND_SOC_AK4641=y CONFIG_SND_SOC_AK4642=y CONFIG_SND_SOC_AK4671=y CONFIG_SND_SOC_ALC5623=y CONFIG_SND_SOC_ALC5632=y CONFIG_SND_SOC_CS42L51=y CONFIG_SND_SOC_CS42L52=y CONFIG_SND_SOC_CS42L73=y CONFIG_SND_SOC_CS4270=y CONFIG_SND_SOC_CS4271=y CONFIG_SND_SOC_CX20442=y CONFIG_SND_SOC_JZ4740_CODEC=y CONFIG_SND_SOC_L3=y CONFIG_SND_SOC_DA7210=y CONFIG_SND_SOC_DA7213=y CONFIG_SND_SOC_DA732X=y CONFIG_SND_SOC_DA9055=y CONFIG_SND_SOC_DFBMCS320=y CONFIG_SND_SOC_ISABELLE=y CONFIG_SND_SOC_LM49453=y CONFIG_SND_SOC_MAX98088=y CONFIG_SND_SOC_MAX98090=y CONFIG_SND_SOC_MAX98095=y CONFIG_SND_SOC_MAX9850=y CONFIG_SND_SOC_PCM3008=y CONFIG_SND_SOC_RT5631=y CONFIG_SND_SOC_SGTL5000=y CONFIG_SND_SOC_SPDIF=y CONFIG_SND_SOC_SSM2602=y CONFIG_SND_SOC_STA32X=y CONFIG_SND_SOC_STA529=y CONFIG_SND_SOC_TLV320AIC23=y CONFIG_SND_SOC_TLV320AIC26=y CONFIG_SND_SOC_TLV320AIC32X4=y CONFIG_SND_SOC_TLV320AIC3X=y CONFIG_SND_SOC_TLV320DAC33=y CONFIG_SND_SOC_TWL4030=y CONFIG_SND_SOC_UDA134X=y CONFIG_SND_SOC_UDA1380=y CONFIG_SND_SOC_WM0010=y CONFIG_SND_SOC_WM1250_EV1=y CONFIG_SND_SOC_WM2000=y CONFIG_SND_SOC_WM2200=y CONFIG_SND_SOC_WM5100=y CONFIG_SND_SOC_WM5102=y CONFIG_SND_SOC_WM8350=y CONFIG_SND_SOC_WM8510=y CONFIG_SND_SOC_WM8523=y CONFIG_SND_SOC_WM8580=y CONFIG_SND_SOC_WM8711=y CONFIG_SND_SOC_WM8727=y CONFIG_SND_SOC_WM8728=y CONFIG_SND_SOC_WM8731=y CONFIG_SND_SOC_WM8737=y CONFIG_SND_SOC_WM8741=y CONFIG_SND_SOC_WM8750=y CONFIG_SND_SOC_WM8753=y CONFIG_SND_SOC_WM8770=y CONFIG_SND_SOC_WM8776=y CONFIG_SND_SOC_WM8782=y CONFIG_SND_SOC_WM8804=y CONFIG_SND_SOC_WM8900=y CONFIG_SND_SOC_WM8903=y CONFIG_SND_SOC_WM8904=y CONFIG_SND_SOC_WM8940=y CONFIG_SND_SOC_WM8955=y CONFIG_SND_SOC_WM8960=y CONFIG_SND_SOC_WM8961=y CONFIG_SND_SOC_WM8962=y CONFIG_SND_SOC_WM8971=y CONFIG_SND_SOC_WM8974=y CONFIG_SND_SOC_WM8978=y CONFIG_SND_SOC_WM8983=y CONFIG_SND_SOC_WM8985=y CONFIG_SND_SOC_WM8988=y CONFIG_SND_SOC_WM8990=y CONFIG_SND_SOC_WM8991=y CONFIG_SND_SOC_WM8993=y CONFIG_SND_SOC_WM8994=y CONFIG_SND_SOC_WM8995=y CONFIG_SND_SOC_WM8996=y CONFIG_SND_SOC_WM9081=y CONFIG_SND_SOC_WM9090=y CONFIG_SND_SOC_LM4857=y CONFIG_SND_SOC_MAX9768=y CONFIG_SND_SOC_MAX9877=y CONFIG_SND_SOC_MC13783=y CONFIG_SND_SOC_ML26124=y CONFIG_SND_SOC_TPA6130A2=y # CONFIG_SND_SIMPLE_CARD is not set CONFIG_SOUND_PRIME=m # CONFIG_SOUND_OSS is not set # # HID support # CONFIG_HID=m # CONFIG_HIDRAW is not set CONFIG_UHID=m CONFIG_HID_GENERIC=m # # Special HID drivers # CONFIG_HID_A4TECH=m CONFIG_HID_ACRUX=m CONFIG_HID_ACRUX_FF=y CONFIG_HID_APPLE=m # CONFIG_HID_AUREAL is not set CONFIG_HID_BELKIN=m # CONFIG_HID_CHERRY is not set CONFIG_HID_CHICONY=m # CONFIG_HID_PRODIKEYS is not set # CONFIG_HID_CYPRESS is not set # CONFIG_HID_DRAGONRISE is not set CONFIG_HID_EMS_FF=m CONFIG_HID_EZKEY=m # CONFIG_HID_HOLTEK is not set # CONFIG_HID_KEYTOUCH is not set # CONFIG_HID_KYE is not set # CONFIG_HID_UCLOGIC is not set CONFIG_HID_WALTOP=m # CONFIG_HID_GYRATION is not set # CONFIG_HID_TWINHAN is not set CONFIG_HID_KENSINGTON=m # CONFIG_HID_LCPOWER is not set CONFIG_HID_LENOVO_TPKBD=m # CONFIG_HID_LOGITECH is not set CONFIG_HID_MICROSOFT=m # CONFIG_HID_MONTEREY is not set CONFIG_HID_MULTITOUCH=m CONFIG_HID_NTRIG=m CONFIG_HID_ORTEK=m # CONFIG_HID_PANTHERLORD is not set CONFIG_HID_PETALYNX=m # CONFIG_HID_PICOLCD is not set CONFIG_HID_PRIMAX=m CONFIG_HID_ROCCAT=m CONFIG_HID_SAITEK=m CONFIG_HID_SAMSUNG=m # CONFIG_HID_SONY is not set CONFIG_HID_SPEEDLINK=m CONFIG_HID_STEELSERIES=m # CONFIG_HID_SUNPLUS is not set CONFIG_HID_GREENASIA=m CONFIG_GREENASIA_FF=y # CONFIG_HID_SMARTJOYPLUS is not set # CONFIG_HID_TIVO is not set CONFIG_HID_TOPSEED=m # CONFIG_HID_THINGM is not set # CONFIG_HID_THRUSTMASTER is not set # CONFIG_HID_ZEROPLUS is not set CONFIG_HID_ZYDACRON=m CONFIG_HID_SENSOR_HUB=m # # USB HID support # CONFIG_USB_HID=m # CONFIG_HID_PID is not set CONFIG_USB_HIDDEV=y # # USB HID Boot Protocol drivers # CONFIG_USB_KBD=m CONFIG_USB_MOUSE=m # # I2C HID support # CONFIG_I2C_HID=m CONFIG_USB_ARCH_HAS_OHCI=y CONFIG_USB_ARCH_HAS_EHCI=y CONFIG_USB_ARCH_HAS_XHCI=y CONFIG_USB_SUPPORT=y CONFIG_USB_COMMON=y CONFIG_USB_ARCH_HAS_HCD=y CONFIG_USB=y # CONFIG_USB_DEBUG is not set # CONFIG_USB_ANNOUNCE_NEW_DEVICES is not set # # Miscellaneous USB options # CONFIG_USB_DYNAMIC_MINORS=y # CONFIG_USB_SUSPEND is not set # CONFIG_USB_OTG_WHITELIST is not set CONFIG_USB_OTG_BLACKLIST_HUB=y CONFIG_USB_DWC3=m # CONFIG_USB_DWC3_HOST is not set CONFIG_USB_DWC3_GADGET=y # CONFIG_USB_DWC3_DUAL_ROLE is not set # CONFIG_USB_DWC3_DEBUG is not set CONFIG_USB_MON=y CONFIG_USB_WUSB_CBAF=m CONFIG_USB_WUSB_CBAF_DEBUG=y # # USB Host Controller Drivers # CONFIG_USB_C67X00_HCD=y CONFIG_USB_XHCI_HCD=y CONFIG_USB_XHCI_PLATFORM=m # CONFIG_USB_XHCI_HCD_DEBUGGING is not set CONFIG_USB_EHCI_HCD=y CONFIG_USB_EHCI_ROOT_HUB_TT=y CONFIG_USB_EHCI_TT_NEWSCHED=y CONFIG_USB_EHCI_PCI=y CONFIG_USB_OXU210HP_HCD=y CONFIG_USB_ISP116X_HCD=m CONFIG_USB_ISP1760_HCD=m # CONFIG_USB_ISP1362_HCD is not set # CONFIG_USB_OHCI_HCD is not set CONFIG_USB_EHCI_HCD_PLATFORM=m CONFIG_USB_UHCI_HCD=y CONFIG_USB_U132_HCD=m CONFIG_USB_SL811_HCD=y # CONFIG_USB_SL811_HCD_ISO is not set # CONFIG_USB_SL811_CS is not set CONFIG_USB_R8A66597_HCD=m # CONFIG_USB_HCD_BCMA is not set CONFIG_USB_HCD_SSB=m CONFIG_USB_MUSB_HDRC=m # CONFIG_USB_MUSB_TUSB6010 is not set CONFIG_MUSB_PIO_ONLY=y CONFIG_USB_CHIPIDEA=y CONFIG_USB_CHIPIDEA_HOST=y # CONFIG_USB_CHIPIDEA_DEBUG is not set # CONFIG_USB_RENESAS_USBHS is not set # # USB Device Class drivers # CONFIG_USB_ACM=y CONFIG_USB_PRINTER=y CONFIG_USB_WDM=y CONFIG_USB_TMC=y # # NOTE: USB_STORAGE depends on SCSI but BLK_DEV_SD may # # # also be needed; see USB_STORAGE Help for more info # # # USB Imaging devices # CONFIG_USB_MDC800=m # # USB port drivers # CONFIG_USB_USS720=y CONFIG_USB_SERIAL=y CONFIG_USB_SERIAL_CONSOLE=y # CONFIG_USB_SERIAL_GENERIC is not set # CONFIG_USB_SERIAL_AIRCABLE is not set CONFIG_USB_SERIAL_ARK3116=m CONFIG_USB_SERIAL_BELKIN=y # CONFIG_USB_SERIAL_CH341 is not set CONFIG_USB_SERIAL_WHITEHEAT=y CONFIG_USB_SERIAL_DIGI_ACCELEPORT=y # CONFIG_USB_SERIAL_CP210X is not set CONFIG_USB_SERIAL_CYPRESS_M8=y # CONFIG_USB_SERIAL_EMPEG is not set # CONFIG_USB_SERIAL_FTDI_SIO is not set # CONFIG_USB_SERIAL_FUNSOFT is not set # CONFIG_USB_SERIAL_VISOR is not set CONFIG_USB_SERIAL_IPAQ=y # CONFIG_USB_SERIAL_IR is not set CONFIG_USB_SERIAL_EDGEPORT=m CONFIG_USB_SERIAL_EDGEPORT_TI=m CONFIG_USB_SERIAL_F81232=y CONFIG_USB_SERIAL_GARMIN=m CONFIG_USB_SERIAL_IPW=y CONFIG_USB_SERIAL_IUU=m CONFIG_USB_SERIAL_KEYSPAN_PDA=m # CONFIG_USB_SERIAL_KEYSPAN is not set CONFIG_USB_SERIAL_KLSI=m CONFIG_USB_SERIAL_KOBIL_SCT=m CONFIG_USB_SERIAL_MCT_U232=m # CONFIG_USB_SERIAL_METRO is not set CONFIG_USB_SERIAL_MOS7720=y CONFIG_USB_SERIAL_MOS7715_PARPORT=y CONFIG_USB_SERIAL_MOS7840=y CONFIG_USB_SERIAL_MOTOROLA=m CONFIG_USB_SERIAL_NAVMAN=y # CONFIG_USB_SERIAL_PL2303 is not set CONFIG_USB_SERIAL_OTI6858=m # CONFIG_USB_SERIAL_QCAUX is not set CONFIG_USB_SERIAL_QUALCOMM=m # CONFIG_USB_SERIAL_SPCP8X5 is not set CONFIG_USB_SERIAL_HP4X=m CONFIG_USB_SERIAL_SAFE=m CONFIG_USB_SERIAL_SAFE_PADDED=y CONFIG_USB_SERIAL_SIEMENS_MPI=m CONFIG_USB_SERIAL_SIERRAWIRELESS=y CONFIG_USB_SERIAL_SYMBOL=m CONFIG_USB_SERIAL_TI=m # CONFIG_USB_SERIAL_CYBERJACK is not set CONFIG_USB_SERIAL_XIRCOM=m CONFIG_USB_SERIAL_WWAN=y # CONFIG_USB_SERIAL_OPTION is not set CONFIG_USB_SERIAL_OMNINET=y CONFIG_USB_SERIAL_OPTICON=m CONFIG_USB_SERIAL_VIVOPAY_SERIAL=y CONFIG_USB_SERIAL_XSENS_MT=m CONFIG_USB_SERIAL_ZIO=m CONFIG_USB_SERIAL_ZTE=y CONFIG_USB_SERIAL_SSU100=y CONFIG_USB_SERIAL_QT2=y CONFIG_USB_SERIAL_DEBUG=y # # USB Miscellaneous drivers # # CONFIG_USB_EMI62 is not set # CONFIG_USB_EMI26 is not set CONFIG_USB_ADUTUX=y CONFIG_USB_SEVSEG=m CONFIG_USB_RIO500=y CONFIG_USB_LEGOTOWER=m # CONFIG_USB_LCD is not set # CONFIG_USB_LED is not set # CONFIG_USB_CYPRESS_CY7C63 is not set CONFIG_USB_CYTHERM=m # CONFIG_USB_IDMOUSE is not set CONFIG_USB_FTDI_ELAN=y CONFIG_USB_APPLEDISPLAY=m # CONFIG_USB_SISUSBVGA is not set # CONFIG_USB_LD is not set # CONFIG_USB_TRANCEVIBRATOR is not set CONFIG_USB_IOWARRIOR=m # CONFIG_USB_TEST is not set CONFIG_USB_ISIGHTFW=m # CONFIG_USB_YUREX is not set CONFIG_USB_EZUSB_FX2=y CONFIG_USB_HSIC_USB3503=m # # USB Physical Layer drivers # CONFIG_OMAP_USB3=y CONFIG_OMAP_CONTROL_USB=y CONFIG_USB_ISP1301=y CONFIG_USB_RCAR_PHY=y CONFIG_USB_GADGET=m CONFIG_USB_GADGET_DEBUG=y CONFIG_USB_GADGET_DEBUG_FS=y CONFIG_USB_GADGET_VBUS_DRAW=2 CONFIG_USB_GADGET_STORAGE_NUM_BUFFERS=2 # # USB Peripheral Controller # CONFIG_USB_R8A66597=m CONFIG_USB_MV_UDC=m # CONFIG_USB_GADGET_MUSB_HDRC is not set CONFIG_USB_M66592=m # CONFIG_USB_AMD5536UDC is not set CONFIG_USB_NET2272=m # CONFIG_USB_NET2272_DMA is not set CONFIG_USB_NET2280=m CONFIG_USB_GOKU=m # CONFIG_USB_EG20T is not set CONFIG_USB_DUMMY_HCD=m CONFIG_USB_LIBCOMPOSITE=m CONFIG_USB_F_ACM=m CONFIG_USB_F_SS_LB=m CONFIG_USB_U_SERIAL=m CONFIG_USB_ZERO=m # CONFIG_USB_AUDIO is not set CONFIG_USB_GADGETFS=m # CONFIG_USB_FUNCTIONFS is not set CONFIG_USB_G_SERIAL=m CONFIG_USB_MIDI_GADGET=m # CONFIG_USB_G_PRINTER is not set # CONFIG_USB_G_HID is not set CONFIG_USB_G_DBGP=m CONFIG_USB_G_DBGP_PRINTK=y # CONFIG_USB_G_DBGP_SERIAL is not set CONFIG_USB_G_WEBCAM=m # # OTG and related infrastructure # CONFIG_USB_OTG_UTILS=y CONFIG_USB_GPIO_VBUS=m # CONFIG_NOP_USB_XCEIV is not set # CONFIG_UWB is not set CONFIG_MMC=m CONFIG_MMC_DEBUG=y CONFIG_MMC_UNSAFE_RESUME=y # CONFIG_MMC_CLKGATE is not set # # MMC/SD/SDIO Card Drivers # # CONFIG_SDIO_UART is not set # CONFIG_MMC_TEST is not set # # MMC/SD/SDIO Host Controller Drivers # CONFIG_MMC_SDHCI=m CONFIG_MMC_SDHCI_PCI=m # CONFIG_MMC_RICOH_MMC is not set # CONFIG_MMC_SDHCI_PLTFM is not set CONFIG_MMC_WBSD=m CONFIG_MMC_TIFM_SD=m CONFIG_MMC_SPI=m # CONFIG_MMC_SDRICOH_CS is not set CONFIG_MMC_CB710=m CONFIG_MMC_VIA_SDMMC=m # CONFIG_MMC_VUB300 is not set CONFIG_MMC_USHC=m CONFIG_MMC_REALTEK_PCI=m CONFIG_MEMSTICK=y # CONFIG_MEMSTICK_DEBUG is not set # # MemoryStick drivers # CONFIG_MEMSTICK_UNSAFE_RESUME=y # # MemoryStick Host Controller Drivers # CONFIG_MEMSTICK_TIFM_MS=y # CONFIG_MEMSTICK_JMICRON_38X is not set # CONFIG_MEMSTICK_R592 is not set CONFIG_MEMSTICK_REALTEK_PCI=m CONFIG_NEW_LEDS=y CONFIG_LEDS_CLASS=y # # LED drivers # # CONFIG_LEDS_LM3530 is not set # CONFIG_LEDS_LM3642 is not set CONFIG_LEDS_PCA9532=m CONFIG_LEDS_PCA9532_GPIO=y CONFIG_LEDS_GPIO=y CONFIG_LEDS_LP3944=y # CONFIG_LEDS_LP5521 is not set CONFIG_LEDS_LP5523=m CONFIG_LEDS_CLEVO_MAIL=m CONFIG_LEDS_PCA955X=y # CONFIG_LEDS_PCA9633 is not set # CONFIG_LEDS_WM831X_STATUS is not set CONFIG_LEDS_WM8350=m # CONFIG_LEDS_DAC124S085 is not set # CONFIG_LEDS_REGULATOR is not set CONFIG_LEDS_BD2802=m CONFIG_LEDS_INTEL_SS4200=m CONFIG_LEDS_LT3593=m CONFIG_LEDS_MC13783=m CONFIG_LEDS_TCA6507=y # CONFIG_LEDS_MAX8997 is not set # CONFIG_LEDS_LM355x is not set CONFIG_LEDS_OT200=y CONFIG_LEDS_BLINKM=y CONFIG_LEDS_TRIGGERS=y # # LED Triggers # CONFIG_LEDS_TRIGGER_TIMER=y CONFIG_LEDS_TRIGGER_ONESHOT=m CONFIG_LEDS_TRIGGER_HEARTBEAT=y # CONFIG_LEDS_TRIGGER_BACKLIGHT is not set CONFIG_LEDS_TRIGGER_CPU=y CONFIG_LEDS_TRIGGER_GPIO=y CONFIG_LEDS_TRIGGER_DEFAULT_ON=m # # iptables trigger is under Netfilter config (LED target) # CONFIG_LEDS_TRIGGER_TRANSIENT=y # CONFIG_ACCESSIBILITY is not set CONFIG_EDAC=y CONFIG_EDAC_LEGACY_SYSFS=y CONFIG_EDAC_DEBUG=y CONFIG_EDAC_MM_EDAC=m CONFIG_EDAC_AMD76X=m CONFIG_EDAC_E7XXX=m CONFIG_EDAC_E752X=m # CONFIG_EDAC_I82875P is not set CONFIG_EDAC_I82975X=m CONFIG_EDAC_I3000=m CONFIG_EDAC_I3200=m CONFIG_EDAC_X38=m CONFIG_EDAC_I5400=m # CONFIG_EDAC_I82860 is not set CONFIG_EDAC_R82600=m # CONFIG_EDAC_I5000 is not set # CONFIG_EDAC_I5100 is not set # CONFIG_EDAC_I7300 is not set CONFIG_RTC_LIB=y CONFIG_RTC_CLASS=y CONFIG_RTC_DEBUG=y # # RTC interfaces # # CONFIG_RTC_INTF_SYSFS is not set CONFIG_RTC_INTF_DEV=y # CONFIG_RTC_INTF_DEV_UIE_EMUL is not set CONFIG_RTC_DRV_TEST=m # # I2C RTC drivers # # CONFIG_RTC_DRV_88PM80X is not set CONFIG_RTC_DRV_DS1307=y # CONFIG_RTC_DRV_DS1374 is not set CONFIG_RTC_DRV_DS1672=y CONFIG_RTC_DRV_DS3232=y CONFIG_RTC_DRV_MAX6900=y CONFIG_RTC_DRV_MAX8998=m CONFIG_RTC_DRV_MAX8997=m # CONFIG_RTC_DRV_MAX77686 is not set # CONFIG_RTC_DRV_RS5C372 is not set # CONFIG_RTC_DRV_ISL1208 is not set CONFIG_RTC_DRV_ISL12022=y CONFIG_RTC_DRV_X1205=m CONFIG_RTC_DRV_PALMAS=m # CONFIG_RTC_DRV_PCF8523 is not set # CONFIG_RTC_DRV_PCF8563 is not set CONFIG_RTC_DRV_PCF8583=m CONFIG_RTC_DRV_M41T80=m CONFIG_RTC_DRV_M41T80_WDT=y # CONFIG_RTC_DRV_BQ32K is not set CONFIG_RTC_DRV_TWL4030=m CONFIG_RTC_DRV_TPS65910=y CONFIG_RTC_DRV_RC5T583=m CONFIG_RTC_DRV_S35390A=y # CONFIG_RTC_DRV_FM3130 is not set # CONFIG_RTC_DRV_RX8581 is not set # CONFIG_RTC_DRV_RX8025 is not set # CONFIG_RTC_DRV_EM3027 is not set # CONFIG_RTC_DRV_RV3029C2 is not set # # SPI RTC drivers # # CONFIG_RTC_DRV_M41T93 is not set # CONFIG_RTC_DRV_M41T94 is not set # CONFIG_RTC_DRV_DS1305 is not set CONFIG_RTC_DRV_DS1390=y CONFIG_RTC_DRV_MAX6902=m # CONFIG_RTC_DRV_R9701 is not set CONFIG_RTC_DRV_RS5C348=y CONFIG_RTC_DRV_DS3234=y CONFIG_RTC_DRV_PCF2123=y CONFIG_RTC_DRV_RX4581=m # # Platform RTC drivers # CONFIG_RTC_DRV_CMOS=y CONFIG_RTC_DRV_DS1286=y CONFIG_RTC_DRV_DS1511=y CONFIG_RTC_DRV_DS1553=m CONFIG_RTC_DRV_DS1742=y CONFIG_RTC_DRV_STK17TA8=m CONFIG_RTC_DRV_M48T86=m CONFIG_RTC_DRV_M48T35=y CONFIG_RTC_DRV_M48T59=y # CONFIG_RTC_DRV_MSM6242 is not set CONFIG_RTC_DRV_BQ4802=y # CONFIG_RTC_DRV_RP5C01 is not set CONFIG_RTC_DRV_V3020=y CONFIG_RTC_DRV_DS2404=m CONFIG_RTC_DRV_WM831X=y # CONFIG_RTC_DRV_WM8350 is not set # CONFIG_RTC_DRV_PCF50633 is not set # # on-CPU RTC drivers # # CONFIG_RTC_DRV_PCAP is not set # CONFIG_RTC_DRV_MC13XXX is not set # # HID Sensor RTC drivers # CONFIG_RTC_DRV_HID_SENSOR_TIME=m # CONFIG_DMADEVICES is not set CONFIG_AUXDISPLAY=y # CONFIG_KS0108 is not set CONFIG_UIO=m # CONFIG_UIO_CIF is not set CONFIG_UIO_PDRV=m CONFIG_UIO_PDRV_GENIRQ=m # CONFIG_UIO_DMEM_GENIRQ is not set CONFIG_UIO_AEC=m CONFIG_UIO_SERCOS3=m CONFIG_UIO_PCI_GENERIC=m CONFIG_UIO_NETX=m CONFIG_VIRTIO=y # # Virtio drivers # CONFIG_VIRTIO_PCI=m # CONFIG_VIRTIO_BALLOON is not set # CONFIG_VIRTIO_MMIO is not set # # Microsoft Hyper-V guest support # CONFIG_STAGING=y CONFIG_ECHO=m CONFIG_COMEDI=m # CONFIG_COMEDI_DEBUG is not set CONFIG_COMEDI_DEFAULT_BUF_SIZE_KB=2048 CONFIG_COMEDI_DEFAULT_BUF_MAXSIZE_KB=20480 CONFIG_COMEDI_MISC_DRIVERS=y CONFIG_COMEDI_KCOMEDILIB=m # CONFIG_COMEDI_BOND is not set CONFIG_COMEDI_TEST=m CONFIG_COMEDI_PARPORT=m # CONFIG_COMEDI_SERIAL2002 is not set CONFIG_COMEDI_SKEL=m CONFIG_COMEDI_PCI_DRIVERS=y CONFIG_COMEDI_8255_PCI=m CONFIG_COMEDI_ADDI_WATCHDOG=m # CONFIG_COMEDI_ADDI_APCI_035 is not set # CONFIG_COMEDI_ADDI_APCI_1032 is not set # CONFIG_COMEDI_ADDI_APCI_1500 is not set # CONFIG_COMEDI_ADDI_APCI_1516 is not set CONFIG_COMEDI_ADDI_APCI_1564=m CONFIG_COMEDI_ADDI_APCI_16XX=m CONFIG_COMEDI_ADDI_APCI_2032=m CONFIG_COMEDI_ADDI_APCI_2200=m CONFIG_COMEDI_ADDI_APCI_3120=m CONFIG_COMEDI_ADDI_APCI_3501=m CONFIG_COMEDI_ADDI_APCI_3XXX=m CONFIG_COMEDI_ADL_PCI6208=m CONFIG_COMEDI_ADL_PCI7X3X=m CONFIG_COMEDI_ADL_PCI8164=m CONFIG_COMEDI_ADL_PCI9111=m # CONFIG_COMEDI_ADL_PCI9118 is not set CONFIG_COMEDI_ADV_PCI1710=m # CONFIG_COMEDI_ADV_PCI1723 is not set CONFIG_COMEDI_ADV_PCI_DIO=m # CONFIG_COMEDI_AMPLC_DIO200_PCI is not set CONFIG_COMEDI_AMPLC_PC236_PCI=m # CONFIG_COMEDI_AMPLC_PC263_PCI is not set CONFIG_COMEDI_AMPLC_PCI224=m # CONFIG_COMEDI_AMPLC_PCI230 is not set # CONFIG_COMEDI_CONTEC_PCI_DIO is not set CONFIG_COMEDI_DAS08_PCI=m # CONFIG_COMEDI_DT3000 is not set CONFIG_COMEDI_DYNA_PCI10XX=m # CONFIG_COMEDI_UNIOXX5 is not set CONFIG_COMEDI_GSC_HPDI=m CONFIG_COMEDI_ICP_MULTI=m CONFIG_COMEDI_II_PCI20KC=m # CONFIG_COMEDI_DAQBOARD2000 is not set CONFIG_COMEDI_JR3_PCI=m CONFIG_COMEDI_KE_COUNTER=m CONFIG_COMEDI_CB_PCIDAS64=m CONFIG_COMEDI_CB_PCIDAS=m # CONFIG_COMEDI_CB_PCIDDA is not set CONFIG_COMEDI_CB_PCIMDAS=m CONFIG_COMEDI_CB_PCIMDDA=m # CONFIG_COMEDI_ME4000 is not set CONFIG_COMEDI_ME_DAQ=m CONFIG_COMEDI_NI_6527=m CONFIG_COMEDI_NI_65XX=m CONFIG_COMEDI_NI_660X=m CONFIG_COMEDI_NI_670X=m # CONFIG_COMEDI_NI_LABPC is not set # CONFIG_COMEDI_NI_PCIDIO is not set CONFIG_COMEDI_NI_PCIMIO=m CONFIG_COMEDI_RTD520=m # CONFIG_COMEDI_S526 is not set CONFIG_COMEDI_S626=m CONFIG_COMEDI_SSV_DNP=m CONFIG_COMEDI_MITE=m CONFIG_COMEDI_NI_TIOCMD=m CONFIG_COMEDI_PCMCIA_DRIVERS=y # CONFIG_COMEDI_CB_DAS16_CS is not set # CONFIG_COMEDI_DAS08_CS is not set # CONFIG_COMEDI_NI_DAQ_700_CS is not set CONFIG_COMEDI_NI_DAQ_DIO24_CS=m CONFIG_COMEDI_NI_MIO_CS=m # CONFIG_COMEDI_QUATECH_DAQP_CS is not set # CONFIG_COMEDI_USB_DRIVERS is not set CONFIG_COMEDI_8255=m CONFIG_COMEDI_FC=m CONFIG_COMEDI_AMPLC_PC236=m CONFIG_COMEDI_DAS08=m CONFIG_COMEDI_NI_TIO=m # CONFIG_ASUS_OLED is not set CONFIG_PANEL=y CONFIG_PANEL_PARPORT=0 CONFIG_PANEL_PROFILE=5 CONFIG_PANEL_CHANGE_MESSAGE=y CONFIG_PANEL_BOOT_MESSAGE="" CONFIG_TRANZPORT=m CONFIG_LINE6_USB=m CONFIG_LINE6_USB_IMPULSE_RESPONSE=y CONFIG_USB_SERIAL_QUATECH2=m CONFIG_DX_SEP=m # # IIO staging drivers # # CONFIG_IIO_ST_HWMON is not set # # Accelerometers # # CONFIG_ADIS16201 is not set CONFIG_ADIS16203=m CONFIG_ADIS16204=m CONFIG_ADIS16209=y CONFIG_ADIS16220=m # CONFIG_ADIS16240 is not set # CONFIG_LIS3L02DQ is not set # CONFIG_SCA3000 is not set # # Analog to digital converters # CONFIG_AD7291=y CONFIG_AD7606=m CONFIG_AD7606_IFACE_PARALLEL=m # CONFIG_AD7606_IFACE_SPI is not set CONFIG_AD799X=m CONFIG_AD799X_RING_BUFFER=y CONFIG_AD7780=m # CONFIG_AD7816 is not set # CONFIG_AD7192 is not set # CONFIG_ADT7410 is not set CONFIG_AD7280=m # # Analog digital bi-direction converters # # CONFIG_ADT7316 is not set # # Capacitance to digital converters # CONFIG_AD7150=m CONFIG_AD7152=y CONFIG_AD7746=y # # Direct Digital Synthesis # # CONFIG_AD5930 is not set CONFIG_AD9832=m CONFIG_AD9834=y # CONFIG_AD9850 is not set # CONFIG_AD9852 is not set CONFIG_AD9910=m # CONFIG_AD9951 is not set # # Digital gyroscope sensors # # CONFIG_ADIS16060 is not set # CONFIG_ADIS16130 is not set # CONFIG_ADIS16260 is not set # # Network Analyzer, Impedance Converters # CONFIG_AD5933=m # # Light sensors # CONFIG_SENSORS_ISL29018=y CONFIG_SENSORS_ISL29028=y CONFIG_TSL2583=y # CONFIG_TSL2x7x is not set # # Magnetometer sensors # # CONFIG_SENSORS_AK8975 is not set CONFIG_SENSORS_HMC5843=y # # Active energy metering IC # # CONFIG_ADE7753 is not set CONFIG_ADE7754=y CONFIG_ADE7758=m # CONFIG_ADE7759 is not set CONFIG_ADE7854=m CONFIG_ADE7854_I2C=m # CONFIG_ADE7854_SPI is not set # # Resolver to digital converters # CONFIG_AD2S90=y CONFIG_AD2S1200=y CONFIG_AD2S1210=y # # Triggers - standalone # CONFIG_IIO_PERIODIC_RTC_TRIGGER=y # CONFIG_IIO_GPIO_TRIGGER is not set # CONFIG_IIO_SYSFS_TRIGGER is not set CONFIG_IIO_DUMMY_EVGEN=y CONFIG_IIO_SIMPLE_DUMMY=y CONFIG_IIO_SIMPLE_DUMMY_EVENTS=y CONFIG_IIO_SIMPLE_DUMMY_BUFFER=y # CONFIG_ZSMALLOC is not set CONFIG_CRYSTALHD=y # CONFIG_FT1000 is not set # # Speakup console speech # CONFIG_TOUCHSCREEN_CLEARPAD_TM1217=m CONFIG_TOUCHSCREEN_SYNAPTICS_I2C_RMI4=m CONFIG_STAGING_MEDIA=y CONFIG_VIDEO_DT3155=m CONFIG_DT3155_CCIR=y # CONFIG_DT3155_STREAMING is not set CONFIG_SOLO6X10=m # # Android # # CONFIG_ANDROID is not set # CONFIG_NET_VENDOR_SILICOM is not set CONFIG_CED1401=m CONFIG_DGRP=m CONFIG_X86_PLATFORM_DEVICES=y CONFIG_DELL_LAPTOP=m CONFIG_SENSORS_HDAPS=m # CONFIG_IBM_RTL is not set # CONFIG_SAMSUNG_LAPTOP is not set CONFIG_SAMSUNG_Q10=m # # Hardware Spinlock drivers # CONFIG_CLKSRC_I8253=y CONFIG_CLKEVT_I8253=y CONFIG_CLKBLD_I8253=y # CONFIG_MAILBOX is not set # CONFIG_IOMMU_SUPPORT is not set # # Remoteproc drivers # CONFIG_REMOTEPROC=y CONFIG_STE_MODEM_RPROC=y # # Rpmsg drivers # CONFIG_VIRT_DRIVERS=y CONFIG_PM_DEVFREQ=y # # DEVFREQ Governors # CONFIG_DEVFREQ_GOV_SIMPLE_ONDEMAND=y CONFIG_DEVFREQ_GOV_PERFORMANCE=y CONFIG_DEVFREQ_GOV_POWERSAVE=y CONFIG_DEVFREQ_GOV_USERSPACE=y # # DEVFREQ Drivers # CONFIG_EXTCON=y # # Extcon Device Drivers # CONFIG_EXTCON_GPIO=y CONFIG_EXTCON_ADC_JACK=m # CONFIG_EXTCON_MAX8997 is not set CONFIG_EXTCON_ARIZONA=m CONFIG_MEMORY=y CONFIG_IIO=y CONFIG_IIO_BUFFER=y # CONFIG_IIO_BUFFER_CB is not set CONFIG_IIO_KFIFO_BUF=y CONFIG_IIO_TRIGGERED_BUFFER=y CONFIG_IIO_TRIGGER=y CONFIG_IIO_CONSUMERS_PER_TRIGGER=2 # # Accelerometers # CONFIG_HID_SENSOR_ACCEL_3D=m CONFIG_KXSD9=y # CONFIG_IIO_ST_ACCEL_3AXIS is not set # # Analog to digital converters # CONFIG_AD_SIGMA_DELTA=m CONFIG_AD7266=m CONFIG_AD7298=m CONFIG_AD7791=m # CONFIG_AD7793 is not set CONFIG_AD7476=m CONFIG_AD7887=y CONFIG_MAX1363=m CONFIG_TI_ADC081C=m CONFIG_TI_AM335X_ADC=m # # Amplifiers # CONFIG_AD8366=m # # Hid Sensor IIO Common # CONFIG_HID_SENSOR_IIO_COMMON=m CONFIG_HID_SENSOR_IIO_TRIGGER=m CONFIG_HID_SENSOR_ENUM_BASE_QUIRKS=y CONFIG_IIO_ST_SENSORS_I2C=y CONFIG_IIO_ST_SENSORS_SPI=y CONFIG_IIO_ST_SENSORS_CORE=y # # Digital to analog converters # CONFIG_AD5064=m # CONFIG_AD5360 is not set CONFIG_AD5380=y # CONFIG_AD5421 is not set CONFIG_AD5624R_SPI=y # CONFIG_AD5446 is not set CONFIG_AD5449=y # CONFIG_AD5504 is not set CONFIG_AD5755=y # CONFIG_AD5764 is not set CONFIG_AD5791=m CONFIG_AD5686=m CONFIG_MAX517=m CONFIG_MCP4725=m # # Frequency Synthesizers DDS/PLL # # # Clock Generator/Distribution # CONFIG_AD9523=m # # Phase-Locked Loop (PLL) frequency synthesizers # CONFIG_ADF4350=y # # Digital gyroscope sensors # # CONFIG_ADIS16080 is not set CONFIG_ADIS16136=m # CONFIG_ADXRS450 is not set # CONFIG_HID_SENSOR_GYRO_3D is not set CONFIG_IIO_ST_GYRO_3AXIS=y CONFIG_IIO_ST_GYRO_I2C_3AXIS=y CONFIG_IIO_ST_GYRO_SPI_3AXIS=y CONFIG_ITG3200=y # # Inertial measurement units # # CONFIG_ADIS16400 is not set # CONFIG_ADIS16480 is not set CONFIG_IIO_ADIS_LIB=y CONFIG_IIO_ADIS_LIB_BUFFER=y # CONFIG_INV_MPU6050_IIO is not set # # Light sensors # # CONFIG_ADJD_S311 is not set # CONFIG_SENSORS_TSL2563 is not set # CONFIG_VCNL4000 is not set CONFIG_HID_SENSOR_ALS=m # # Magnetometer sensors # # CONFIG_HID_SENSOR_MAGNETOMETER_3D is not set CONFIG_IIO_ST_MAGN_3AXIS=m CONFIG_IIO_ST_MAGN_I2C_3AXIS=m CONFIG_IIO_ST_MAGN_SPI_3AXIS=m CONFIG_VME_BUS=y # # VME Bridge Drivers # # CONFIG_VME_CA91CX42 is not set # CONFIG_VME_TSI148 is not set # # VME Board Drivers # # CONFIG_VMIVME_7805 is not set # # VME Device Drivers # CONFIG_VME_USER=y CONFIG_VME_PIO2=m CONFIG_PWM=y CONFIG_PWM_TWL=m CONFIG_PWM_TWL_LED=m CONFIG_IPACK_BUS=m CONFIG_BOARD_TPCI200=m # CONFIG_SERIAL_IPOCTAL is not set # # Firmware Drivers # CONFIG_EDD=m CONFIG_EDD_OFF=y CONFIG_FIRMWARE_MEMMAP=y CONFIG_DELL_RBU=y CONFIG_DCDBAS=m # CONFIG_DMIID is not set CONFIG_DMI_SYSFS=y # CONFIG_ISCSI_IBFT_FIND is not set CONFIG_GOOGLE_FIRMWARE=y # # Google Firmware Drivers # # CONFIG_GOOGLE_MEMCONSOLE is not set # # File systems # CONFIG_DCACHE_WORD_ACCESS=y # CONFIG_FS_POSIX_ACL is not set CONFIG_FILE_LOCKING=y # CONFIG_FSNOTIFY is not set # CONFIG_DNOTIFY is not set # CONFIG_INOTIFY_USER is not set # CONFIG_FANOTIFY is not set # CONFIG_QUOTA is not set # CONFIG_QUOTACTL is not set CONFIG_AUTOFS4_FS=m CONFIG_FUSE_FS=m # CONFIG_CUSE is not set # # Caches # CONFIG_FSCACHE=y # CONFIG_FSCACHE_DEBUG is not set # # Pseudo filesystems # # CONFIG_PROC_FS is not set CONFIG_SYSFS=y # CONFIG_TMPFS is not set # CONFIG_HUGETLBFS is not set # CONFIG_HUGETLB_PAGE is not set CONFIG_CONFIGFS_FS=y CONFIG_MISC_FILESYSTEMS=y CONFIG_ECRYPT_FS=y # CONFIG_PSTORE is not set CONFIG_NLS=y CONFIG_NLS_DEFAULT="iso8859-1" CONFIG_NLS_CODEPAGE_437=m # CONFIG_NLS_CODEPAGE_737 is not set CONFIG_NLS_CODEPAGE_775=m CONFIG_NLS_CODEPAGE_850=m CONFIG_NLS_CODEPAGE_852=y CONFIG_NLS_CODEPAGE_855=y # CONFIG_NLS_CODEPAGE_857 is not set CONFIG_NLS_CODEPAGE_860=y CONFIG_NLS_CODEPAGE_861=y # CONFIG_NLS_CODEPAGE_862 is not set CONFIG_NLS_CODEPAGE_863=m CONFIG_NLS_CODEPAGE_864=y CONFIG_NLS_CODEPAGE_865=m CONFIG_NLS_CODEPAGE_866=m CONFIG_NLS_CODEPAGE_869=y CONFIG_NLS_CODEPAGE_936=m CONFIG_NLS_CODEPAGE_950=m # CONFIG_NLS_CODEPAGE_932 is not set # CONFIG_NLS_CODEPAGE_949 is not set # CONFIG_NLS_CODEPAGE_874 is not set # CONFIG_NLS_ISO8859_8 is not set CONFIG_NLS_CODEPAGE_1250=m CONFIG_NLS_CODEPAGE_1251=m CONFIG_NLS_ASCII=y # CONFIG_NLS_ISO8859_1 is not set CONFIG_NLS_ISO8859_2=y CONFIG_NLS_ISO8859_3=y # CONFIG_NLS_ISO8859_4 is not set CONFIG_NLS_ISO8859_5=m # CONFIG_NLS_ISO8859_6 is not set CONFIG_NLS_ISO8859_7=m CONFIG_NLS_ISO8859_9=m CONFIG_NLS_ISO8859_13=m CONFIG_NLS_ISO8859_14=y CONFIG_NLS_ISO8859_15=y # CONFIG_NLS_KOI8_R is not set # CONFIG_NLS_KOI8_U is not set CONFIG_NLS_MAC_ROMAN=m CONFIG_NLS_MAC_CELTIC=y CONFIG_NLS_MAC_CENTEURO=y CONFIG_NLS_MAC_CROATIAN=y CONFIG_NLS_MAC_CYRILLIC=y # CONFIG_NLS_MAC_GAELIC is not set # CONFIG_NLS_MAC_GREEK is not set # CONFIG_NLS_MAC_ICELAND is not set # CONFIG_NLS_MAC_INUIT is not set CONFIG_NLS_MAC_ROMANIAN=y # CONFIG_NLS_MAC_TURKISH is not set CONFIG_NLS_UTF8=y # # Kernel hacking # CONFIG_TRACE_IRQFLAGS_SUPPORT=y CONFIG_PRINTK_TIME=y CONFIG_DEFAULT_MESSAGE_LOGLEVEL=4 # CONFIG_ENABLE_WARN_DEPRECATED is not set # CONFIG_ENABLE_MUST_CHECK is not set CONFIG_FRAME_WARN=1024 CONFIG_MAGIC_SYSRQ=y # CONFIG_STRIP_ASM_SYMS is not set # CONFIG_READABLE_ASM is not set CONFIG_UNUSED_SYMBOLS=y CONFIG_DEBUG_FS=y CONFIG_HEADERS_CHECK=y # CONFIG_DEBUG_SECTION_MISMATCH is not set CONFIG_DEBUG_KERNEL=y CONFIG_DEBUG_SHIRQ=y # CONFIG_LOCKUP_DETECTOR is not set CONFIG_PANIC_ON_OOPS=y CONFIG_PANIC_ON_OOPS_VALUE=1 # CONFIG_DETECT_HUNG_TASK is not set # CONFIG_DEBUG_OBJECTS is not set # CONFIG_DEBUG_SLAB is not set CONFIG_HAVE_DEBUG_KMEMLEAK=y # CONFIG_DEBUG_KMEMLEAK is not set CONFIG_DEBUG_PREEMPT=y CONFIG_DEBUG_RT_MUTEXES=y CONFIG_DEBUG_PI_LIST=y # CONFIG_RT_MUTEX_TESTER is not set CONFIG_DEBUG_SPINLOCK=y CONFIG_DEBUG_MUTEXES=y CONFIG_DEBUG_LOCK_ALLOC=y # CONFIG_PROVE_LOCKING is not set CONFIG_LOCKDEP=y CONFIG_LOCK_STAT=y CONFIG_DEBUG_LOCKDEP=y CONFIG_DEBUG_ATOMIC_SLEEP=y # CONFIG_DEBUG_LOCKING_API_SELFTESTS is not set CONFIG_STACKTRACE=y # CONFIG_DEBUG_STACK_USAGE is not set # CONFIG_DEBUG_KOBJECT is not set # CONFIG_DEBUG_BUGVERBOSE is not set # CONFIG_DEBUG_INFO is not set CONFIG_DEBUG_VM=y # CONFIG_DEBUG_VM_RB is not set CONFIG_DEBUG_VIRTUAL=y # CONFIG_DEBUG_WRITECOUNT is not set # CONFIG_DEBUG_MEMORY_INIT is not set CONFIG_DEBUG_LIST=y # CONFIG_TEST_LIST_SORT is not set CONFIG_DEBUG_SG=y # CONFIG_DEBUG_NOTIFIERS is not set CONFIG_DEBUG_CREDENTIALS=y CONFIG_ARCH_WANT_FRAME_POINTERS=y CONFIG_FRAME_POINTER=y CONFIG_BOOT_PRINTK_DELAY=y # # RCU Debugging # # CONFIG_PROVE_RCU_DELAY is not set CONFIG_SPARSE_RCU_POINTER=y # CONFIG_RCU_TORTURE_TEST is not set CONFIG_RCU_CPU_STALL_TIMEOUT=21 CONFIG_RCU_CPU_STALL_VERBOSE=y CONFIG_RCU_CPU_STALL_INFO=y # CONFIG_RCU_TRACE is not set # CONFIG_BACKTRACE_SELF_TEST is not set # CONFIG_DEBUG_FORCE_WEAK_PER_CPU is not set # CONFIG_NOTIFIER_ERROR_INJECTION is not set CONFIG_FAULT_INJECTION=y CONFIG_FAILSLAB=y CONFIG_FAIL_PAGE_ALLOC=y # CONFIG_FAIL_MMC_REQUEST is not set # CONFIG_FAULT_INJECTION_DEBUG_FS is not set # CONFIG_DEBUG_PAGEALLOC is not set CONFIG_USER_STACKTRACE_SUPPORT=y CONFIG_NOP_TRACER=y CONFIG_HAVE_FUNCTION_TRACER=y CONFIG_HAVE_FUNCTION_GRAPH_TRACER=y CONFIG_HAVE_FUNCTION_GRAPH_FP_TEST=y CONFIG_HAVE_FUNCTION_TRACE_MCOUNT_TEST=y CONFIG_HAVE_DYNAMIC_FTRACE=y CONFIG_HAVE_DYNAMIC_FTRACE_WITH_REGS=y CONFIG_HAVE_FTRACE_MCOUNT_RECORD=y CONFIG_HAVE_SYSCALL_TRACEPOINTS=y CONFIG_HAVE_C_RECORDMCOUNT=y CONFIG_TRACER_MAX_TRACE=y CONFIG_TRACE_CLOCK=y CONFIG_RING_BUFFER=y CONFIG_EVENT_TRACING=y CONFIG_CONTEXT_SWITCH_TRACER=y CONFIG_RING_BUFFER_ALLOW_SWAP=y CONFIG_TRACING=y CONFIG_GENERIC_TRACER=y CONFIG_TRACING_SUPPORT=y CONFIG_FTRACE=y CONFIG_FUNCTION_TRACER=y # CONFIG_IRQSOFF_TRACER is not set CONFIG_PREEMPT_TRACER=y CONFIG_SCHED_TRACER=y CONFIG_FTRACE_SYSCALLS=y CONFIG_TRACER_SNAPSHOT=y CONFIG_TRACE_BRANCH_PROFILING=y # CONFIG_BRANCH_PROFILE_NONE is not set # CONFIG_PROFILE_ANNOTATED_BRANCHES is not set CONFIG_PROFILE_ALL_BRANCHES=y # CONFIG_BRANCH_TRACER is not set CONFIG_STACK_TRACER=y CONFIG_UPROBE_EVENT=y CONFIG_PROBE_EVENTS=y CONFIG_DYNAMIC_FTRACE=y CONFIG_DYNAMIC_FTRACE_WITH_REGS=y CONFIG_FUNCTION_PROFILER=y CONFIG_FTRACE_MCOUNT_RECORD=y CONFIG_FTRACE_SELFTEST=y CONFIG_FTRACE_STARTUP_TEST=y # CONFIG_EVENT_TRACE_TEST_SYSCALLS is not set CONFIG_MMIOTRACE=y CONFIG_MMIOTRACE_TEST=m CONFIG_RING_BUFFER_BENCHMARK=y CONFIG_RBTREE_TEST=m CONFIG_INTERVAL_TREE_TEST=m CONFIG_PROVIDE_OHCI1394_DMA_INIT=y CONFIG_BUILD_DOCSRC=y # CONFIG_DYNAMIC_DEBUG is not set # CONFIG_DMA_API_DEBUG is not set # CONFIG_ATOMIC64_SELFTEST is not set # CONFIG_SAMPLES is not set CONFIG_HAVE_ARCH_KGDB=y CONFIG_KGDB=y CONFIG_KGDB_SERIAL_CONSOLE=y CONFIG_KGDB_TESTS=y CONFIG_KGDB_TESTS_ON_BOOT=y CONFIG_KGDB_TESTS_BOOT_STRING="V1F100" # CONFIG_KGDB_LOW_LEVEL_TRAP is not set # CONFIG_KGDB_KDB is not set CONFIG_HAVE_ARCH_KMEMCHECK=y # CONFIG_TEST_KSTRTOX is not set # CONFIG_STRICT_DEVMEM is not set # CONFIG_X86_VERBOSE_BOOTUP is not set CONFIG_EARLY_PRINTK=y CONFIG_EARLY_PRINTK_DBGP=y CONFIG_DEBUG_STACKOVERFLOW=y # CONFIG_X86_PTDUMP is not set # CONFIG_DEBUG_RODATA is not set # CONFIG_DEBUG_SET_MODULE_RONX is not set # CONFIG_DEBUG_NX_TEST is not set CONFIG_DOUBLEFAULT=y CONFIG_IOMMU_STRESS=y CONFIG_HAVE_MMIOTRACE_SUPPORT=y CONFIG_IO_DELAY_TYPE_0X80=0 CONFIG_IO_DELAY_TYPE_0XED=1 CONFIG_IO_DELAY_TYPE_UDELAY=2 CONFIG_IO_DELAY_TYPE_NONE=3 # CONFIG_IO_DELAY_0X80 is not set CONFIG_IO_DELAY_0XED=y # CONFIG_IO_DELAY_UDELAY is not set # CONFIG_IO_DELAY_NONE is not set CONFIG_DEFAULT_IO_DELAY_TYPE=1 CONFIG_DEBUG_BOOT_PARAMS=y # CONFIG_CPA_DEBUG is not set CONFIG_OPTIMIZE_INLINING=y CONFIG_DEBUG_NMI_SELFTEST=y # # Security options # CONFIG_KEYS=y # CONFIG_TRUSTED_KEYS is not set CONFIG_ENCRYPTED_KEYS=y # CONFIG_KEYS_DEBUG_PROC_KEYS is not set # CONFIG_SECURITY_DMESG_RESTRICT is not set CONFIG_SECURITY=y CONFIG_SECURITYFS=y CONFIG_SECURITY_NETWORK=y CONFIG_SECURITY_PATH=y # CONFIG_SECURITY_YAMA is not set CONFIG_INTEGRITY=y # CONFIG_INTEGRITY_SIGNATURE is not set CONFIG_IMA=y CONFIG_IMA_MEASURE_PCR_IDX=10 # CONFIG_IMA_APPRAISE is not set CONFIG_EVM=y CONFIG_EVM_HMAC_VERSION=2 CONFIG_DEFAULT_SECURITY_DAC=y CONFIG_DEFAULT_SECURITY="" CONFIG_CRYPTO=y # # Crypto core or helper # CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_ALGAPI2=y CONFIG_CRYPTO_AEAD=y CONFIG_CRYPTO_AEAD2=y CONFIG_CRYPTO_BLKCIPHER=y CONFIG_CRYPTO_BLKCIPHER2=y CONFIG_CRYPTO_HASH=y CONFIG_CRYPTO_HASH2=y CONFIG_CRYPTO_RNG=y CONFIG_CRYPTO_RNG2=y CONFIG_CRYPTO_PCOMP2=y CONFIG_CRYPTO_MANAGER=y CONFIG_CRYPTO_MANAGER2=y CONFIG_CRYPTO_MANAGER_DISABLE_TESTS=y CONFIG_CRYPTO_GF128MUL=y CONFIG_CRYPTO_NULL=m CONFIG_CRYPTO_WORKQUEUE=y CONFIG_CRYPTO_CRYPTD=y CONFIG_CRYPTO_AUTHENC=y CONFIG_CRYPTO_TEST=m CONFIG_CRYPTO_ABLK_HELPER_X86=y CONFIG_CRYPTO_GLUE_HELPER_X86=m # # Authenticated Encryption with Associated Data # # CONFIG_CRYPTO_CCM is not set CONFIG_CRYPTO_GCM=m CONFIG_CRYPTO_SEQIV=y # # Block modes # CONFIG_CRYPTO_CBC=y CONFIG_CRYPTO_CTR=m CONFIG_CRYPTO_CTS=m CONFIG_CRYPTO_ECB=y CONFIG_CRYPTO_LRW=y CONFIG_CRYPTO_PCBC=y CONFIG_CRYPTO_XTS=y # # Hash modes # CONFIG_CRYPTO_HMAC=y # CONFIG_CRYPTO_XCBC is not set # CONFIG_CRYPTO_VMAC is not set # # Digest # CONFIG_CRYPTO_CRC32C=y # CONFIG_CRYPTO_CRC32C_INTEL is not set CONFIG_CRYPTO_GHASH=m CONFIG_CRYPTO_MD4=y CONFIG_CRYPTO_MD5=y CONFIG_CRYPTO_MICHAEL_MIC=m CONFIG_CRYPTO_RMD128=y # CONFIG_CRYPTO_RMD160 is not set CONFIG_CRYPTO_RMD256=m # CONFIG_CRYPTO_RMD320 is not set CONFIG_CRYPTO_SHA1=y CONFIG_CRYPTO_SHA256=y CONFIG_CRYPTO_SHA512=y CONFIG_CRYPTO_TGR192=y # CONFIG_CRYPTO_WP512 is not set # # Ciphers # CONFIG_CRYPTO_AES=y CONFIG_CRYPTO_AES_586=y CONFIG_CRYPTO_AES_NI_INTEL=y # CONFIG_CRYPTO_ANUBIS is not set CONFIG_CRYPTO_ARC4=m CONFIG_CRYPTO_BLOWFISH=y CONFIG_CRYPTO_BLOWFISH_COMMON=y CONFIG_CRYPTO_CAMELLIA=m CONFIG_CRYPTO_CAST_COMMON=y CONFIG_CRYPTO_CAST5=y CONFIG_CRYPTO_CAST6=m CONFIG_CRYPTO_DES=y # CONFIG_CRYPTO_FCRYPT is not set CONFIG_CRYPTO_KHAZAD=m CONFIG_CRYPTO_SALSA20=y CONFIG_CRYPTO_SALSA20_586=y CONFIG_CRYPTO_SEED=m CONFIG_CRYPTO_SERPENT=m CONFIG_CRYPTO_SERPENT_SSE2_586=m CONFIG_CRYPTO_TEA=y CONFIG_CRYPTO_TWOFISH=m CONFIG_CRYPTO_TWOFISH_COMMON=m CONFIG_CRYPTO_TWOFISH_586=m # # Compression # CONFIG_CRYPTO_DEFLATE=m # CONFIG_CRYPTO_ZLIB is not set CONFIG_CRYPTO_LZO=m # # Random Number Generation # # CONFIG_CRYPTO_ANSI_CPRNG is not set CONFIG_CRYPTO_HW=y CONFIG_CRYPTO_DEV_PADLOCK=m # CONFIG_CRYPTO_DEV_PADLOCK_AES is not set # CONFIG_CRYPTO_DEV_PADLOCK_SHA is not set CONFIG_CRYPTO_DEV_GEODE=m CONFIG_CRYPTO_DEV_HIFN_795X=y # CONFIG_CRYPTO_DEV_HIFN_795X_RNG is not set CONFIG_ASYMMETRIC_KEY_TYPE=y CONFIG_ASYMMETRIC_PUBLIC_KEY_SUBTYPE=y CONFIG_PUBLIC_KEY_ALGO_RSA=y CONFIG_X509_CERTIFICATE_PARSER=y CONFIG_HAVE_KVM=y CONFIG_VIRTUALIZATION=y CONFIG_BINARY_PRINTF=y # # Library routines # CONFIG_BITREVERSE=y CONFIG_GENERIC_STRNCPY_FROM_USER=y CONFIG_GENERIC_STRNLEN_USER=y CONFIG_GENERIC_FIND_FIRST_BIT=y CONFIG_GENERIC_PCI_IOMAP=y CONFIG_GENERIC_IOMAP=y CONFIG_GENERIC_IO=y CONFIG_PERCPU_RWSEM=y CONFIG_CRC_CCITT=y CONFIG_CRC16=y CONFIG_CRC_T10DIF=m CONFIG_CRC_ITU_T=y CONFIG_CRC32=y # CONFIG_CRC32_SELFTEST is not set # CONFIG_CRC32_SLICEBY8 is not set # CONFIG_CRC32_SLICEBY4 is not set # CONFIG_CRC32_SARWATE is not set CONFIG_CRC32_BIT=y CONFIG_CRC7=y CONFIG_LIBCRC32C=y # CONFIG_CRC8 is not set CONFIG_ZLIB_INFLATE=m CONFIG_ZLIB_DEFLATE=m CONFIG_LZO_COMPRESS=y CONFIG_LZO_DECOMPRESS=y CONFIG_XZ_DEC=m # CONFIG_XZ_DEC_X86 is not set # CONFIG_XZ_DEC_POWERPC is not set # CONFIG_XZ_DEC_IA64 is not set CONFIG_XZ_DEC_ARM=y # CONFIG_XZ_DEC_ARMTHUMB is not set CONFIG_XZ_DEC_SPARC=y CONFIG_XZ_DEC_BCJ=y CONFIG_XZ_DEC_TEST=m CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y CONFIG_CHECK_SIGNATURE=y CONFIG_ARCH_HAS_ATOMIC64_DEC_IF_POSITIVE=y # CONFIG_AVERAGE is not set CONFIG_CLZ_TAB=y # CONFIG_CORDIC is not set # CONFIG_DDR is not set CONFIG_MPILIB=y CONFIG_OID_REGISTRY=y

1 0

[alsa-devel] Fwd: Re: Alsa USB Audio regression
by Ulrich Günther 19 Jul '13

19 Jul '13

-------- Original-Nachricht -------- Betreff: Re: [alsa-devel] Alsa USB Audio regression Datum: Tue, 16 Jul 2013 18:27:10 +0200 Von: Ulrich Günther <ulrich.guenther(a)gmx.at> An: Raymond Yau <superquad.vortex2(a)gmail.com> Am 2013-07-16 04:14, schrieb Raymond Yau: > > > >>> > >>> After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't > work at > >>> all, Ubuntu 11.10 and newer have the bug), the headset only plays > sound > >>> at maximum volume level if the volume is turned down just a little no > >>> sound is being played. > >>> > >>> I have filled a Ubuntu bug for 12.04 LTS at: > >>> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . > >>> I was advised to contact this mailing list. > >>> > >>> AlsaInfo for Ubuntu 12.04 LTS with the bug: > >>> > http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 > . > >>> AlsaInfo for Ubuntu 10.04 LTS without the bug: > >>> > http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc > >> > >> > >> > >> It is strange that dB range of your usb audio is 0 to 0.14 (only > small gain and no attenuation) > >> > >> In 10.04 LTS pulseaudio provide software volume control > >> > >> > >> ( 1.095| 0.000) I: alsa-sink.c: Driver does not support hardware > volume control, falling back to software volume control. > >> > >> > >> control.6 { > >> comment.access 'read write' > >> comment.type INTEGER > >> comment.count 1 comment.range '0 - 37' > >> comment.dbmin 0 comment.dbmax 14 > >> iface MIXER > >> name 'Speaker Playback Volume' value 30 } > >> > >> > >> In 12.04 pulseaudio did not provide software volume attenuation > and use the speaker playback volume control > > > > > > So does that mean that the bug is in pulseaudio, or could it be even > deeper like in the kernel or something? > > Are there any significant change in volume when you use "alsamixer > -c0" to change speaker playback volume from min 0dB to 0.14dB ? > > Did pulseaudio provide software attuentation -1.61dB when the range of > hardware volume control is only 0.14dB ? > > * index: 0 > name: <alsa_output.usb-0d8c_USB_Audio-00-Audio.analog-surround-71> > driver: <module-alsa-card.c> > flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY > DYNAMIC_LATENCY > state: SUSPENDED > suspend cause: IDLE > priority: 9049 > volume: 0: 94% 1: 94% 2: 94% 3: 94% 4: 94% 5: 94% 6: 94% 7: 94% > 0: -1,61 dB 1: -1,61 dB 2: -1,61 dB 3: -1,61 dB 4: -1,61 dB 5: > -1,61 dB 6: -1,61 dB 7: -1,61 dB > balance 0,00 > base volume: 99% > -0,14 dB > volume steps: 65537 > muted: no > > It is strange that 5.1 headset only support 8 channels playback > Okay i did try "alsamixer -c0" in Ubuntu 12.04 LTS, i selected the Usb Headset, Speaker was at 0 using the remote control from the headset i could bring the value to 86, so if the volume is turned down the value goes immideatly to 0. With the alsamixer i was able to regulate the value between 0 and 100, db-change is 0,01 at lowest and 0,14 at highest. I could increment/decrement the db-change in steps of 0,01 (with the up/down keys from the keyboard), the volume from the headset-speakers follows accordingly. When using anything besides the "alsamixer -c0" the volume can only be 0 or 86. I was able to play the test sounds for: front center, front left, front right, rear left,rear right and subwoofer every thing came from the right direction, as expected from a 5.1 headset. I was really surprised, thank you for pointing this out, it really didn't occur to me, maybe i saw the alsamixer some years ago but i really couldn't remember that it was there.

3 3

[alsa-devel] [PATCH] ASoC: jack: Use power efficient workqueue
by Mark Brown 18 Jul '13

18 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..7aa26b5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Fix error path
by Fabio Estevam 18 Jul '13

18 Jul '13

If the 'failed to find codec platform device' error path is executed, it should jump to 'fail' label instead of returning an error immediately. 'fail' label will then free the ssi_np and codec_np previously acquired nodes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a9..1d70e27 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -217,7 +217,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + ret = -EINVAL; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: tegra: depend on ARCH_TEGRA, not ARCH_TEGRA_*
by Stephen Warren 18 Jul '13

18 Jul '13

From: Stephen Warren <swarren(a)nvidia.com> I'm planning to remove CONFIG_ARCH_TEGRA_*_SOC, leaving just ARCH_TEGRA. This will reduce the number of configurations that need to be supported by core Tegra code, e.g. CPU idle, hotplug. As a precursor, we need to update Kconfig for all Tegra drivers not to reference ARCH_TEGRA_*_SOC. Stop selecting I2S/AC97/SPDIF controller drivers from the machine driver config options; this doesn't really work if we don't know which specific SoC(s) we're building for. However, set their default values based on SND_SOC_TEGRA, since most people will probably want to enable support for all SoCs. This also avoids the need to change any defconfig files. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Kconfig | 36 +++++++++++++++--------------------- 1 file changed, 15 insertions(+), 21 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 68fd6d0..321a6c9 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -7,8 +7,9 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA20_AC97 - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 AC97 controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -18,15 +19,16 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA20_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 I2C controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -34,9 +36,9 @@ config SND_SOC_TEGRA20_I2S machine drivers to support below. config SND_SOC_TEGRA20_SPDIF - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC - default m + tristate "Tegra20 SPDIF controller" + default SND_SOC_TEGRA + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -44,15 +46,16 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 AHUB module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA30_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 I2S controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -62,8 +65,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -72,8 +73,6 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -82,8 +81,6 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -92,8 +89,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB - select SND_SOC_TEGRA20_AC97 + depends on SND_SOC_TEGRA && GPIOLIB select SND_SOC_WM9712 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -102,7 +98,6 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the @@ -111,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the -- 1.8.1.5

2 8

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i8
by Chris J Arges 18 Jul '13

18 Jul '13

This patch is similar to commit 1762a59d8e8b5e99f6f4a0f292b40f3cacb108ba. The same non-standard interface descriptor causes snd_usb_create_mixer() to fail for the Focusrite Scarlett 18i8 as well. Signed-off-by: Chris J Arges <christopherarges(a)gmail.com> --- sound/usb/quirks-table.h | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7f1585b..b954ec5 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3232,6 +3232,52 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Focusrite Scarlett 18i8 + * + * Avoid mixer creation similar to the Scarlett 18i6 interface. + */ + USB_DEVICE(0x1235, 0x8014), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i8", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, { /* -- 1.7.9.5

3 3

[alsa-devel] [PATCH] ASoC: pcm: Use the power efficient workqueue for delayed powerdown
by Mark Brown 18 Jul '13

18 Jul '13

From: Mark Brown <broonie(a)linaro.org> There is no need to use a normal per-CPU workqueue for delayed power downs as they're not timing or performance critical and waking up a core for them would defeat some of the point. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c6403..f4f68cb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -411,8 +411,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: wm8904: fix the typo error for LINER Mux
by Bo Shen 18 Jul '13

18 Jul '13

fix the typo error, from "LINEL Mux" to "LINER Mux" Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c..bdd575a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1017,7 +1017,7 @@ static const struct soc_enum liner_enum = SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); static const struct snd_kcontrol_new liner_mux = - SOC_DAPM_ENUM("LINEL Mux", liner_enum); + SOC_DAPM_ENUM("LINER Mux", liner_enum); static const char *sidetone_text[] = { "None", "Left", "Right" -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hdmi - poll eld at resume time
by Wang Xingchao 18 Jul '13

18 Jul '13

Hdmi driver may not receive intrinsic event from gfx side when it's in runtime suspend mode. There's no ELD info when exit from runtime suspend. This patch avoid missing ELD info. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..cb8ac66 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1868,12 +1868,33 @@ static void generic_hdmi_free(struct hda_codec *codec) kfree(spec); } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + generic_hdmi_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hdmi_present_sense(per_pin, 1); + } + return 0; +} +#endif + static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, .free = generic_hdmi_free, .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; -- 1.8.1.2

5 29

[alsa-devel] [PATCH, RESEND] ALSA: ASoC: au1x: fix ac97c module build
by Manuel Lauss 17 Jul '13

17 Jul '13

The recent conversion to module_platform_driver left a stray ampersand in there, preventing a build. Cc: Mark Brown <broonie(a)linaro.org> Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Against 3.11-rc1, added Mark to Cc. sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- 1.8.3.2

1 0

[alsa-devel] Sound Card driver for audio codec, mux and dsp.
by Diego Sueiro 17 Jul '13

17 Jul '13

Hi there, I need to develop a solution to expose to userspace a single sound card with mixer controls for: Audio codec : sgtl5000 Audio mux: used to select an audio source on sgtl5000 LINE_IN. Audio dsp :sta309a How can I implement this drivers? Should I have to create a sound-card driver which is going to register and control the other drivers (codec, mux, dsp)? Is there some device driver or tutorial guide that I can use as an example? I'm using the kernel 3.0.15 from Freescale [1]. [1] - http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/log/?h=imx_3.0.… Kind Regards, -- *dS Diego Sueiro

1 1

[alsa-devel] idea: a reserve alsa plugin
by David Henningsson 17 Jul '13

17 Jul '13

Just had an idea which I'll write down here before I forget it again...and I'm not saying I'll implement this anytime soon either, but here goes: There is a device reserve protocol between PulseAudio and JACK2 - when JACK needs the sound card, it'll send a dbus message to PulseAudio and grab a name in D-Bus. However, there are plenty of applications who like to access ALSA directly, without going through JACK2 or PulseAudio. By making a "reserve" plugin, we could have this functionality for those apps too. In practice, if the app usually opens "plughw:0" or "hw:0", it could instead open "reserve:plughw:0" or "reserve:hw:0" to also reserve the device from PulseAudio usage while the device is open. Meanwhile, PulseAudio is free to use other audio devices (which is not the case when using e g pasuspender). How does that sound? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

6 12

[alsa-devel] Help in understanding SND_PCM_TSTAMP_ENABLE mode snd_pcm_htimestamp function
by divya v 17 Jul '13

17 Jul '13

Hi, I am looking for a method to maintain timestamps for every sample read or written. I noticed the *SND_PCM_TSTAMP_ENABLE mode. But the description mentions - "" *pdate timestamp at every hardware position update" Doe this mean that the when snd_pcm_htimestamp() is issued, we will get the last hardware update timestamp or does it mean that it will fetch the timestamp of that particular sample? Thanks, Divya

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl: fsl_ssi: Use devm_ functions
by Fabio Estevam 17 Jul '13

17 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_ functions can make the code cleaner and smaller. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 38 ++++++++++---------------------------- 1 file changed, 10 insertions(+), 28 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..ab8d462 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -680,7 +680,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* The DAI name is the last part of the full name of the node. */ p = strrchr(np->full_name, '/') + 1; - ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), + ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private) + strlen(p), GFP_KERNEL); if (!ssi_private) { dev_err(&pdev->dev, "could not allocate DAI object\n"); @@ -698,26 +698,24 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - goto error_kmalloc; + return ret; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - ret = -ENOMEM; - goto error_kmalloc; + return -ENOMEM; } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); if (ssi_private->irq == NO_IRQ) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - ret = -ENXIO; - goto error_iomap; + return -ENXIO; } /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, - ssi_private); + ret = devm_request_irq(&pdev->dev, ssi_private->irq, fsl_ssi_isr, 0, + ssi_private->name, ssi_private); if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); goto error_irqmap; @@ -739,11 +737,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) u32 dma_events[2]; ssi_private->ssi_on_imx = true; - ssi_private->clk = clk_get(&pdev->dev, NULL); + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); - goto error_irq; + goto error_irqmap; } clk_prepare_enable(ssi_private->clk); @@ -794,7 +792,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error_irq; + goto error_clk; } /* Register with ASoC */ @@ -857,23 +855,12 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } - -error_irq: - free_irq(ssi_private->irq, ssi_private); error_irqmap: irq_dispose_mapping(ssi_private->irq); -error_iomap: - iounmap(ssi_private->ssi); - -error_kmalloc: - kfree(ssi_private); - return ret; } @@ -886,15 +873,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) { imx_pcm_dma_exit(pdev); clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); } snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - - free_irq(ssi_private->irq, ssi_private); irq_dispose_mapping(ssi_private->irq); - - kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); return 0; -- 1.8.1.2

2 2

[alsa-devel] Connecting two Codecs to the same CPU DAI
by Daniel Mack 17 Jul '13

17 Jul '13

Hi Mark, Liam, I'm trying to find a sane solution for systems that have multiple Codecs connected to the same CPU DAI. An example would be a codec that receives the same I2S stream as a S/PDIF transceiver, and which is in fact connected to the very same bus in hardware. Currently, I'm setting up two DAI links in such cases, and userspace has to open both subdevices and issue hw_params calls to each of them in order to set up the hardware correctly. This is cumbersome of course, and I wonder how the core could be augmented in order to handle such cases correctly. Ideally, it should only expose one PCM interface and handle a list of Codecs in the back. Or am I unaware of an existing feature? Thanks, Daniel

2 7

[alsa-devel] [BUG] 3.10.[01] modprobe snd-... hangs
by Philipp Matthias Hahn 17 Jul '13

17 Jul '13

Hello, My x86_64 systems has some trouble loading some ALSA snd-* modules since the upgrade to 3.10.[01]: Several automatic modprobe calls hang, but loading snd-intel-hda and snd-audio-usb by hand still works. # ps axf 460 ? S 0:00 \_ [kworker/u8:3] 1087 ? S 0:00 | \_ [kworker/u8:3] 1092 ? S 0:00 | \_ /sbin/modprobe -q -- snd-seq-client-14 ... 1071 ? S 0:00 sh -c /sbin/modprobe --ignore-install snd && { /sbin/modprobe --quiet snd 1080 ? D 0:00 \_ /sbin/modprobe --quiet snd-seq 1115 ? S 0:00 sh -c /sbin/modprobe --ignore-install snd-rawmidi && { /sbin/modprobe --q 1154 ? S 0:00 \_ /sbin/modprobe --quiet snd-seq-midi 1119 ? S 0:00 sh -c /sbin/modprobe --ignore-install snd-rawmidi && { /sbin/modprobe --q 1148 ? S 0:00 \_ /sbin/modprobe --quiet snd-seq-midi In /var/log/daemon.log I find the following messages: Jul 15 19:58:44 scout udevd[912]: timeout: killing '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc01ip00in00' [1044] Jul 15 19:58:56 scout udevd[912]: '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc01ip00in00' [1044] terminated by signal 9 (Killed) Jul 15 19:58:56 scout udevd[919]: timeout: killing '/sbin/modprobe -b pci:v000010DEd0000026Csv00001462sd00007350bc04sc03i00' [1051] Jul 15 19:58:56 scout udevd[914]: timeout: killing '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc03ip00in01' [1045] Jul 15 19:58:56 scout udevd[919]: '/sbin/modprobe -b pci:v000010DEd0000026Csv00001462sd00007350bc04sc03i00' [1051] terminated by signal 9 (Killed) Jul 15 19:58:56 scout udevd[914]: '/sbin/modprobe -b usb:vFC08p0101d0250dc00dsc00dp00ic01isc03ip00in01' [1045] terminated by signal 9 (Killed) This is a Debian sid system # modprobe -V kmod version 9 # echo t >/proc/sysrq-trigger produces too much output to include the modprobe processes. I already stopped most processes, but that is not enough. # cat /proc/1092/syscall 175 0x7effcad92000 0xafe8 0x7effca9c2f88 0x7effca6ea3aa 0x0 0x0 0x7fff3e3de648 0x7effca6eef8a # cat /proc/1080/syscall 175 0x7fe01b25e000 0xafe8 0x7fe01ae8ef88 0x7fe01abb63aa 0x3 0x0 0x7fffba8cc748 0x7fe01abbaf8a # cat /proc/1154/syscall 175 0x7fc157a7b000 0xafe8 0x7fc1576abf88 0x7fc1573d33aa 0x3 0x0 0x7fff05809e58 0x7fc1573d7f8a # cat /proc/1148/syscall 175 0x7fc7159d5000 0xafe8 0x7fc715605f88 0x7fc71532d3aa 0x3 0x0 0x7fff47a1f3c8 0x7fc715331f8a Is this a know bug or has someone seen similar problems? # cat /proc/modules ip6table_filter 1575 0 - Live 0xffffffffa0893000 ip6_tables 13784 1 ip6table_filter, Live 0xffffffffa088b000 ebtable_nat 1887 0 - Live 0xffffffffa0887000 ebtables 19917 1 ebtable_nat, Live 0xffffffffa087e000 snd_usb_audio 124196 0 - Live 0xffffffffa0851000 snd_usbmidi_lib 16982 1 snd_usb_audio, Live 0xffffffffa0848000 rfcomm 28731 10 - Live 0xffffffffa083a000 bnep 9786 2 - Live 0xffffffffa0833000 snd_hda_codec_realtek 37669 1 - Live 0xffffffffa0823000 snd_hda_intel 31432 0 - Live 0xffffffffa0814000 snd_hda_codec 140669 2 snd_hda_codec_realtek,snd_hda_intel, Live 0xffffffffa07e2000 snd_hwdep 5627 2 snd_usb_audio,snd_hda_codec, Live 0xffffffffa07dd000 iptable_filter 1618 1 - Live 0xffffffffa07d9000 xt_owner 1326 1 - Live 0xffffffffa07d5000 ip_tables 13390 1 iptable_filter, Live 0xffffffffa07cd000 x_tables 18433 6 ip6table_filter,ip6_tables,ebtables,iptable_filter,xt_owner,ip_tables, Live 0xffffffffa07c3000 cpufreq_conservative 7015 0 - Live 0xffffffffa07bd000 cpufreq_ondemand 8945 0 - Live 0xffffffffa07b6000 cpufreq_powersave 1010 0 - Live 0xffffffffa07b2000 cpufreq_userspace 2136 0 - Live 0xffffffffa07ae000 binfmt_misc 6479 1 - Live 0xffffffffa07a9000 bridge 74378 0 - Live 0xffffffffa078d000 stp 1908 1 bridge, Live 0xffffffffa0789000 llc 4485 2 bridge,stp, Live 0xffffffffa0784000 deflate 1873 0 - Live 0xffffffffa0780000 ctr 3633 0 - Live 0xffffffffa077c000 twofish_generic 5715 0 - Live 0xffffffffa0777000 twofish_x86_64_3way 19885 0 - Live 0xffffffffa076f000 twofish_x86_64 5503 1 twofish_x86_64_3way, Live 0xffffffffa076a000 twofish_common 13133 3 twofish_generic,twofish_x86_64_3way,twofish_x86_64, Live 0xffffffffa0763000 camellia_generic 18308 0 - Live 0xffffffffa075b000 camellia_x86_64 45885 0 - Live 0xffffffffa074b000 serpent_sse2_x86_64 44963 0 - Live 0xffffffffa073d000 serpent_generic 21243 1 serpent_sse2_x86_64, Live 0xffffffffa0734000 ablk_helper 2284 1 serpent_sse2_x86_64, Live 0xffffffffa0730000 cryptd 7992 1 ablk_helper, Live 0xffffffffa072a000 xts 2958 3 twofish_x86_64_3way,camellia_x86_64,serpent_sse2_x86_64, Live 0xffffffffa0726000 lrw 3584 3 twofish_x86_64_3way,camellia_x86_64,serpent_sse2_x86_64, Live 0xffffffffa0722000 gf128mul 6580 2 xts,lrw, Live 0xffffffffa071d000 glue_helper 4810 3 twofish_x86_64_3way,camellia_x86_64,serpent_sse2_x86_64, Live 0xffffffffa0718000 blowfish_generic 3194 0 - Live 0xffffffffa0714000 blowfish_x86_64 14084 0 - Live 0xffffffffa070d000 blowfish_common 6425 2 blowfish_generic,blowfish_x86_64, Live 0xffffffffa0708000 cast5_generic 10378 0 - Live 0xffffffffa0702000 cast_common 5351 1 cast5_generic, Live 0xffffffffa06fd000 des_generic 16147 0 - Live 0xffffffffa06f6000 cmac 2460 0 - Live 0xffffffffa06f2000 xcbc 2423 0 - Live 0xffffffffa06ee000 rmd160 7552 0 - Live 0xffffffffa06e9000 nfsd 208660 2 - Live 0xffffffffa06a8000 auth_rpcgss 39179 1 nfsd, Live 0xffffffffa0698000 oid_registry 2490 1 auth_rpcgss, Live 0xffffffffa0694000 nfs_acl 2285 1 nfsd, Live 0xffffffffa0690000 sha512_generic 5023 0 - Live 0xffffffffa068b000 nfs 127233 0 - Live 0xffffffffa065b000 lockd 57962 2 nfsd,nfs, Live 0xffffffffa0645000 fscache 34198 1 nfs, Live 0xffffffffa0636000 sha1_ssse3 9266 0 - Live 0xffffffffa0630000 sha1_generic 2064 1 sha1_ssse3, Live 0xffffffffa062c000 sunrpc 185043 6 nfsd,auth_rpcgss,nfs_acl,nfs,lockd, Live 0xffffffffa05ea000 md5 1803 0 - Live 0xffffffffa05e6000 crypto_null 2624 0 - Live 0xffffffffa05e2000 af_key 25880 0 - Live 0xffffffffa05d6000 xfrm_algo 6101 1 af_key, Live 0xffffffffa05d1000 ves1820 5159 1 - Live 0xffffffffa05cc000 btusb 14810 0 - Live 0xffffffffa05bc000 bluetooth 183572 24 rfcomm,bnep,btusb, Live 0xffffffffa054a000 stv0297 6493 1 - Live 0xffffffffa052f000 snd_rawmidi 17913 1 snd_usbmidi_lib, Live 0xffffffffa0525000 snd_pcm_oss 32083 0 - Live 0xffffffffa0518000 snd_mixer_oss 12485 1 snd_pcm_oss, Live 0xffffffffa0510000 snd_pcm 73594 4 snd_usb_audio,snd_hda_intel,snd_hda_codec,snd_pcm_oss, Live 0xffffffffa04f5000 snd_page_alloc 7172 2 snd_hda_intel,snd_pcm, Live 0xffffffffa04f0000 dvb_ttpci 93699 0 - Live 0xffffffffa04d1000 saa7146_vv 26567 1 dvb_ttpci, Live 0xffffffffa04c5000 videobuf_dma_sg 8015 1 saa7146_vv, Live 0xffffffffa04bf000 videobuf_core 15578 2 saa7146_vv,videobuf_dma_sg, Live 0xffffffffa04b7000 v4l2_common 7907 1 saa7146_vv, Live 0xffffffffa04b1000 budget_ci 21012 0 - Live 0xffffffffa04a7000 videodev 97317 2 saa7146_vv,v4l2_common, Live 0xffffffffa0486000 budget_core 7679 1 budget_ci, Live 0xffffffffa0481000 ttpci_eeprom 1985 2 dvb_ttpci,budget_core, Live 0xffffffffa0479000 saa7146 11356 4 dvb_ttpci,saa7146_vv,budget_ci,budget_core, Live 0xffffffffa046b000 rc_core 13478 2 budget_ci, Live 0xffffffffa044a000 dvb_core 68839 3 dvb_ttpci,budget_ci,budget_core, Live 0xffffffffa0431000 snd_seq_oss 33717 1 - Loading 0xffffffffa041b000 snd_seq_midi_event 5740 1 snd_seq_oss, Live 0xffffffffa0416000 snd_seq 44232 2 snd_seq_oss,snd_seq_midi_event, Live 0xffffffffa0404000 snd_seq_device 5560 3 snd_rawmidi,snd_seq_oss,snd_seq, Live 0xffffffffa03ff000 snd_timer 17417 2 snd_pcm,snd_seq, Live 0xffffffffa03f6000 snd 58793 14 snd_usb_audio,snd_usbmidi_lib,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_rawmidi,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_seq_device,snd_timer, Live 0xffffffffa03df000 pcspkr 1926 0 - Live 0xffffffffa03db000 psmouse 52563 0 - Live 0xffffffffa03c6000 serio_raw 4517 0 - Live 0xffffffffa03c1000 soundcore 5811 1 snd, Live 0xffffffffa03b7000 evdev 8465 0 - Live 0xffffffffa039d000 kvm_intel 122108 0 - Live 0xffffffffa0351000 kvm 342589 1 kvm_intel, Live 0xffffffffa0271000 usb_storage 46166 0 - Live 0xffffffffa025c000 fuse 65800 1 - Live 0xffffffffa0243000 ecryptfs 80646 0 - Live 0xffffffffa0227000 cbc 2574 0 - Live 0xffffffffa0223000 sha256_ssse3 16172 2 - Live 0xffffffffa021c000 sha256_generic 9882 1 sha256_ssse3, Live 0xffffffffa0216000 hmac 2673 1 - Live 0xffffffffa0212000 encrypted_keys 9378 1 ecryptfs, Live 0xffffffffa020b000 autofs4 20984 2 - Live 0xffffffffa0200000 ext4 434295 5 - Live 0xffffffffa0173000 crc16 1541 2 bluetooth,ext4, Live 0xffffffffa016f000 jbd2 83146 1 ext4, Live 0xffffffffa0150000 btrfs 682625 1 - Live 0xffffffffa008b000 crc32c 1610 1 - Live 0xffffffffa0087000 libcrc32c 1147 1 btrfs, Live 0xffffffffa0083000 xor 11190 1 btrfs, Live 0xffffffffa007d000 zlib_deflate 18739 2 deflate,btrfs, Live 0xffffffffa0075000 raid6_pq 92000 1 btrfs, Live 0xffffffffa005a000 nbd 8786 0 - Live 0xffffffffa0053000 firewire_ohci 27516 0 - Live 0xffffffffa0048000 firewire_core 49041 1 firewire_ohci, Live 0xffffffffa0035000 crc_itu_t 1565 1 firewire_core, Live 0xffffffffa0031000 forcedeth 50647 0 - Live 0xffffffffa001e000 floppy 53437 0 - Live 0xffffffffa0009000 ohci_hcd 18001 0 - Live 0xffffffffa0000000 BYtE Philipp -- / / (_)__ __ ____ __ Philipp Hahn / /__/ / _ \/ // /\ \/ / /____/_/_//_/\_,_/ /_/\_\ pmhahn(a)pmhahn.de

7 10

[alsa-devel] [PATCH v2] ASoC: sgtl5000: defer the probe if clock is not found
by Shawn Guo 16 Jul '13

16 Jul '13

It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the regression on imx28 since commit 9e13f34 (ASoC: sgtl5000: Let the codec acquire its clock). [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 [ 1.945906] mxs-sgtl5000 sound.13: ASoC: CODEC (null) not registered [ 1.953787] mxs-sgtl5000 sound.13: snd_soc_register_card failed (-517) [ 1.960865] platform sound.13: Driver mxs-sgtl5000 requests probe deferral Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v1: * Not use ternary operator. Mark, I'm still not going to fix this regression during -rc at clock core level with the risk of breaking clock API users and being yelled by clock maintainer. Shawn sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..f5ec35b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,6 +1527,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + /* Defer the probe to see if the clk will be provided later */ + if (ret == -ENOENT) + return -EPROBE_DEFER; return ret; } -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: tegra: fix compile warning in AC'97 driver
by Stephen Warren 16 Jul '13

16 Jul '13

From: Stephen Warren <swarren(a)nvidia.com> This fixes the following by deleting dead code: sound/soc/tegra/tegra20_ac97.c: In function ‘tegra20_ac97_platform_probe’: sound/soc/tegra/tegra20_ac97.c:435:1: warning: label ‘err_unregister_pcm’ defined but not used [-Wunused-label] Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra20_ac97.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e58233f..87b845f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -432,8 +432,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) return 0; -err_unregister_pcm: - tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_asoc_utils_fini: -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoc: tegra: Remove unneeded mach-type.h incldues
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra_alc5632.c | 2 -- sound/soc/tegra/tegra_wm8753.c | 2 -- sound/soc/tegra/trimslice.c | 2 -- 3 files changed, 6 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9..c61ea3a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -13,8 +13,6 @@ * published by the Free Software Foundation. */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f87fc53..8e774d1 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -28,8 +28,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 05c68aa..734bfcd 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -24,8 +24,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: tegra_alc5632: Remove unused mach-types.h include
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra_alc5632.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9..c61ea3a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -13,8 +13,6 @@ * published by the Free Software Foundation. */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> -- 1.8.3.2

2 2

[alsa-devel] [PATCH 0/10] snd_pcm_stop() lock fixes
by Takashi Iwai 16 Jul '13

16 Jul '13

Hi, while reviewing another patch series, I stumbled on the snd_pcm_lock() call in an unlocked context. Then I started looking through the whole tree, and found a bunch of drivers doing it wrong, too. So here is a patch series to fix them. [PATCH 01/10] ALSA: asihpi: Fix unlocked snd_pcm_stop() call [PATCH 02/10] ALSA: atiixp: Fix unlocked snd_pcm_stop() call [PATCH 03/10] ALSA: 6fire: Fix unlocked snd_pcm_stop() call [PATCH 04/10] ALSA: ua101: Fix unlocked snd_pcm_stop() call [PATCH 05/10] ALSA: usx2y: Fix unlocked snd_pcm_stop() call [PATCH 06/10] ALSA: pxa2xx: Fix unlocked snd_pcm_stop() call [PATCH 07/10] ASoC: atmel: Fix unlocked snd_pcm_stop() call [PATCH 08/10] ASoC: s6000: Fix unlocked snd_pcm_stop() call [PATCH 09/10] [media] saa7134: Fix unlocked snd_pcm_stop() call [PATCH 10/10] staging: line6: Fix unlocked snd_pcm_stop() call Takashi

6 22

[alsa-devel] snd_pcm_delay issue - How to debug?
by Sander Jansen 16 Jul '13

16 Jul '13

Hi, I'm seeing an issue where snd_pcm_delay is returning bogus values (usually a lot larger than in actuality) in that it ultimately affects the progress timer in my audio player. Both are HDA based audio cards, though one connects to the HDMI output and while the other connects to analog out. I'm suspecting I'm seeing a driver bug since I only start noticing the issue since a few kernel releases ago (though not sure which one started it). How can i best debug this issue and find out the cause. Thanks, Sander

1 0

[alsa-devel] [PATCH] ASoC: imx-sgtl5000: fix error return code in imx_sgtl5000_probe()
by Wei Yongjun 16 Jul '13

16 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return a negative error code from the error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/imx-sgtl5000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 7a8bc12..25f310d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,8 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); goto fail; + } data->clk_frequency = clk_get_rate(data->codec_clk);

3 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Add compatible strings for specific devices
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> The driver supports a range of devices but currently doesn't allow those device names to be used for enumeration on DT. Add the currently listed I2C IDs as compatible strings. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 8 +++++++- sound/soc/codecs/tlv320aic3x.c | 2 ++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index f47c3f5..26f65f9 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -3,7 +3,13 @@ Texas Instruments - tlv320aic3x Codec module The tlv320aic3x serial control bus communicates through I2C protocols Required properties: -- compatible - "string" - "ti,tlv320aic3x" + +- compatible - "string" - One of: + "ti,tlv320aic3x" - Generic TLV320AIC3x device + "ti,tlv320aic33" - TLV320AIC33 + "ti,tlv320aic3007" - TLV320AIC3007 + + - reg - <int> - I2C slave address diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b9268..c9bb760 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1582,6 +1582,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) #if defined(CONFIG_OF) static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, + { .compatible = "ti,tlv320aic33" }, + { .compatible = "ti,tlv320aic3007" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- 1.8.3.2

1 1

[alsa-devel] [PATCH] ASoC: imx: Enable COMPILE_TEST builds
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa43854..87e28bc 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -98,7 +98,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. -- 1.8.3.2

2 3

[alsa-devel] [PATCH v2] ASoC: kirkwood-i2s: fix a compilation warning
by Jean-Francois Moine 16 Jul '13

16 Jul '13

In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function The ALSA subsystem should never gives such a rate because: - the rates with the internal clock are limited to 44.1, 48 and 96 kHz as specified by the kirkwood_i2s_dai structure, - the other rates are proposed in the structure kirkwood_i2s_dai_extclk only when the external clock is present. In case of programming error (bad rate for internal clock and no external clock), the function will simply cause a backtrace. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 - add more explanations --- sound/soc/kirkwood/kirkwood-i2s.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..c9daf3b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -105,14 +105,16 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for supported rates */ + /* use internal dco for the supported rates + * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { - /* use optional external clk for other rates */ + } else { + /* use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate);

2 1

[alsa-devel] [RFC 1/2] ASoC: dmaengine-pcm: Add support for querying DMA capabilities
by Lars-Peter Clausen 16 Jul '13

16 Jul '13

Currently each platform making use the the generic dmaengine PCM driver still needs to provide a custom snd_pcm_hardware struct which specifies the capabilities of the DMA controller, e.g. the maximum period size that can be supported. This patch adds code which uses the newly introduced dma_get_slave_caps() API to query this information from the dmaengine driver. The new code path will only be taken if the 'pcm_hardware' field of the snd_dmaengine_pcm_config struct is NULL. The patch also introduces a new 'fifo_size' field to the snd_dmaengine_dai_dma_data struct which is used to initialize the snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- For review only for now. The dma_get_slave_caps() API has not been merged yet, the patch adding support for it can be found here: http://permalink.gmane.org/gmane.linux.kernel/1524880 --- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-generic-dmaengine-pcm.c | 46 +++++++++++++++++++++++++++++++++-- 2 files changed, 46 insertions(+), 2 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35c..83b2c3e 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. + * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -68,6 +69,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + unsigned int fifo_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3c..98044a7 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -92,6 +92,43 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } +static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data; + struct snd_pcm_hardware hw; + struct dma_slave_caps dma_caps; + int ret; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret) + return ret; + + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = dma_caps.max_sg_nr; + hw.period_bytes_min = 16; + hw.period_bytes_max = dma_caps.max_sg_len; + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = dma_data->fifo_size; + + if (hw.periods_max == 0) + hw.periods_max = UINT_MAX; + if (hw.period_bytes_max == 0) + hw.period_bytes_max = SIZE_MAX; + + if (dma_caps.cmd_pause) + hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + + return snd_soc_set_runtime_hwparams(substream, &hw); +} + static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -99,8 +136,13 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream) struct dma_chan *chan = pcm->chan[substream->stream]; int ret; - ret = snd_soc_set_runtime_hwparams(substream, - pcm->config->pcm_hardware); + if (pcm->config->pcm_hardware) { + ret = snd_soc_set_runtime_hwparams(substream, + pcm->config->pcm_hardware); + } else { + ret = dmaengine_pcm_set_runtime_hwparams(substream); + } + if (ret) return ret; -- 1.8.0

3 8

[alsa-devel] [PATCH] ALSA: hda - Headphone mic support for an Asus/Conexant device
by David Henningsson 16 Jul '13

16 Jul '13

This Conexant codec has a single jack that can be used as either headphone or mic (but not headset). The existing hp_mic functionality does not apply here, because the mic and the HP are on separate pins. Hence make a lighter version of what has been earlier done for Realtek codecs. BugLink: https://bugs.launchpad.net/bugs/1198030 Tested-by: Franz Hsieh <franz.hsieh(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 79 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 78 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index de00ce1..4edd2d0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -66,6 +66,8 @@ struct conexant_spec { hda_nid_t eapds[4]; bool dynamic_eapd; + unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + #ifdef ENABLE_CXT_STATIC_QUIRKS const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec) snd_hda_gen_init(codec); if (!spec->dynamic_eapd) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + return 0; } @@ -3224,6 +3229,8 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_HEADPHONE_MIC_PIN, + CXT_FIXUP_HEADPHONE_MIC, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec, (0 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_update_headset_mode(struct hda_codec *codec) +{ + /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */ + int i; + bool mic_mode = false; + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == mux_pin) { + mic_mode = !!cfg->inputs[i].is_headphone_mic; + break; + } + + if (mic_mode) { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */ + spec->gen.hp_jack_present = false; + } else { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */ + spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]); + } + + snd_hda_gen_update_outputs(codec); +} + +static void cxt_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + cxt_update_headset_mode(codec); +} + +static void cxt_fixup_headphone_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; + spec->gen.automute_hook = cxt_update_headset_mode; + break; + case HDA_FIXUP_ACT_INIT: + cxt_update_headset_mode(codec); + break; + } +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_HEADPHONE_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADPHONE_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { } + } + }, + [CXT_FIXUP_HEADPHONE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headphone_mic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), @@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, + spec->parse_flags); if (err < 0) goto error; @@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->bus->allow_bus_reset = 1; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: kirkwood-i2s: Remove empty remove()
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/kirkwood/kirkwood-i2s.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 44412ea..becc082 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -398,11 +398,6 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) } -static int kirkwood_i2s_remove(struct snd_soc_dai *dai) -{ - return 0; -} - static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .startup = kirkwood_i2s_startup, .trigger = kirkwood_i2s_trigger, @@ -413,7 +408,6 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { static struct snd_soc_dai_driver kirkwood_i2s_dai = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, @@ -431,7 +425,6 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, -- 1.8.3.2

1 1

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID to snd-hda
by Aaron Plattner 16 Jul '13

16 Jul '13

Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip. Reviewed-by: Andy Ritger <aritger(a)nvidia.com> Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..486def7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2536,6 +2536,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2588,6 +2589,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- 1.7.12

3 3

Re: [alsa-devel] Alsa USB Audio regression
by Raymond Yau 16 Jul '13

16 Jul '13

>>> >>> After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't work at >>> all, Ubuntu 11.10 and newer have the bug), the headset only plays sound >>> at maximum volume level if the volume is turned down just a little no >>> sound is being played. >>> >>> I have filled a Ubuntu bug for 12.04 LTS at: >>> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . >>> I was advised to contact this mailing list. >>> >>> AlsaInfo for Ubuntu 12.04 LTS with the bug: >>> http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 . >>> AlsaInfo for Ubuntu 10.04 LTS without the bug: >>> http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc >> >> >> >> It is strange that dB range of your usb audio is 0 to 0.14 (only small gain and no attenuation) >> >> In 10.04 LTS pulseaudio provide software volume control >> >> >> ( 1.095| 0.000) I: alsa-sink.c: Driver does not support hardware volume control, falling back to software volume control. >> >> >> control.6 { >> comment.access 'read write' >> comment.type INTEGER >> comment.count 1 comment.range '0 - 37' >> comment.dbmin 0 comment.dbmax 14 >> iface MIXER >> name 'Speaker Playback Volume' value 30 } >> >> >> In 12.04 pulseaudio did not provide software volume attenuation and use the speaker playback volume control > > > So does that mean that the bug is in pulseaudio, or could it be even deeper like in the kernel or something? Are there any significant change in volume when you use "alsamixer -c0" to change speaker playback volume from min 0dB to 0.14dB ? Did pulseaudio provide software attuentation -1.61dB when the range of hardware volume control is only 0.14dB ? * index: 0 name: <alsa_output.usb-0d8c_USB_Audio-00-Audio.analog-surround-71> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE priority: 9049 volume: 0: 94% 1: 94% 2: 94% 3: 94% 4: 94% 5: 94% 6: 94% 7: 94% 0: -1,61 dB 1: -1,61 dB 2: -1,61 dB 3: -1,61 dB 4: -1,61 dB 5: -1,61 dB 6: -1,61 dB 7: -1,61 dB balance 0,00 base volume: 99% -0,14 dB volume steps: 65537 muted: no It is strange that 5.1 headset only support 8 channels playback

1 0

[alsa-devel] [PATCH] ASoC: kirkwood-i2s: fix a compilation warning
by Jean-Francois Moine 16 Jul '13

16 Jul '13

In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function As the ALSA subsystem should never gives such a rate, this patch removes the check of the external clock pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..8da5cdb 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -111,7 +111,7 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { + } else { /* use optional external clk for other rates */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate);

3 7

[alsa-devel] [PATCH] ALSA: hda - Fix error accessing non-existent node for Realtek codecs
by David Henningsson 15 Jul '13

15 Jul '13

A recent commit caused my hda-emu based test tool to complain about accessing nodes that don't exist. This simple test works around this. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) I don't have ALC5505 hardware here, no alsa-info to look at, and I don't know what it is. Therefore it would be good with some confirmation that this patch does not break such hardware, which could be the case if the Realtek codec breaks the HDA spec by having "hidden" nodes. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..ca73f10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3844,7 +3844,8 @@ static int patch_alc269(struct hda_codec *codec) break; } - if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { + if (codec->num_nodes > 0x51 && + snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { spec->has_alc5505_dsp = true; spec->init_hook = alc5505_dsp_init; } -- 1.7.9.5

3 5

[alsa-devel] [PATCH RESEND] ASoC: max98088 - fix element type of the register cache.
by Dylan Reid 15 Jul '13

15 Jul '13

From: Chih-Chung Chang <chihchung(a)chromium.org> The registers of max98088 are 8 bits, not 16 bits. This bug causes the contents of registers to be overwritten with bad values when the codec is suspended and then resumed. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada5..566a367 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) -- 1.8.1.3.605.g02339dd

3 2

[alsa-devel] [PATCH] ASoC: sgtl5000: defer the probe if clock is not found
by Shawn Guo 15 Jul '13

15 Jul '13

It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the following problem on imx28. [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Mark, This is a fix for 3.11. Shawn sound/soc/codecs/sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7ebf923 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,7 +1527,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); - return ret; + /* Defer the probe to see if the clk will be provided later */ + return ret == -ENOENT ? -EPROBE_DEFER : ret; } ret = clk_prepare_enable(sgtl5000->mclk); -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: omap: Enable COMPILE_TEST build for DT platforms
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> The DT platforms don't have any source dependency on any OMAP stuff so allow them to be built when COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/omap/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 9f5d55e..68dfc0a 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on ARCH_OMAP && DMA_OMAP + depends on (ARCH_OMAP && DMA_OMAP) || COMPILE_TEST select SND_SOC_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + depends on SND_OMAP_SOC && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 @@ -87,7 +87,7 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- 1.8.3.2

3 5

Re: [alsa-devel] [PATCH 1/1] ASoC: max98088 - fix element type of the register cache.
by Dylan Reid 15 Jul '13

15 Jul '13

+Maintainers and Maxim, Chih-Chung, please CC maintainers on patches. On Tue, Jun 25, 2013 at 12:47 AM, <me(a)google.com> wrote: > From: Chih-Chung Chang <chihchung(a)chromium.org> > > The registers of max98088 are 8 bits, not 16 bits. This bug causes the > contents of registers to be overwritten with bad values when the codec > is suspended and then resumed. > > Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> > > diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c > index 3eeada5..566a367 100644 > --- a/sound/soc/codecs/max98088.c > +++ b/sound/soc/codecs/max98088.c > @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) > > static void max98088_sync_cache(struct snd_soc_codec *codec) > { > - u16 *reg_cache = codec->reg_cache; > + u8 *reg_cache = codec->reg_cache; > int i; > > if (!codec->cache_sync) > -- > 1.8.3 >

2 1

[alsa-devel] [PATCH] ASoC: mxs: Enable COMPILE_TEST builds
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mxs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 78d321c..7daf860 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,6 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" - depends on ARCH_MXS + depends on ARCH_MXS || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

1 0

[alsa-devel] [PATCH 1/3] ASoC: kirkwood: Remove unused headers
by Mark Brown 15 Jul '13

15 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/kirkwood/kirkwood-openrd.c | 2 -- sound/soc/kirkwood/kirkwood-t5325.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..addbebc 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -16,9 +16,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/cs42l51.h" static int openrd_client_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..4f4cb56 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -15,9 +15,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <mach/kirkwood.h> #include <linux/platform_data/asoc-kirkwood.h> -#include <asm/mach-types.h> #include "../codecs/alc5623.h" static int t5325_hw_params(struct snd_pcm_substream *substream, -- 1.8.3.2

1 2

[alsa-devel] [PATCH v5 0/7] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 15 Jul '13

15 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130711 [I let the original signed-off from Nicolas and Uwe in place, I don't know if I should replace them by something like "originaly-signed-off-by" since the code has been changed.] Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards Richard Genoud (6): sound: codec: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 41 ++++ arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 24 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 207 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 57 +++++- 8 files changed, 363 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

2 12

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Let the codec acquire its clock
by Fabio Estevam 15 Jul '13

15 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx6qsabrelite board the following error happens on probe: sgtl5000: probe of 0-000a failed with error -5 imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer Prior to reading the codec ID we need to turn the SYS_MCLK clock, so let's enable the codec clock inside sgtl5000_i2c_probe(). Also remove the codec clock enable/disable functions from the machine driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Shawn, I haven't had a chance to convert mx28 to pass the codec clock via device tree yet. It looks like it is a bit trickier there, as it uses the saif clock. Given that mx28 audio playback is already broken in linux-next, this series does not make things worse on mx28 side. sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/fsl/imx-sgtl5000.c | 30 +++++++----------------------- 2 files changed, 37 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8f2afb..2e0227b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -114,6 +114,7 @@ struct sgtl5000_priv { struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; struct regmap *regmap; + struct clk *mclk; }; /* @@ -1522,16 +1523,28 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + sgtl5000->mclk = devm_clk_get(&client->dev, NULL); + if (IS_ERR(sgtl5000->mclk)) { + ret = PTR_ERR(sgtl5000->mclk); + dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + return ret; + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) - return ret; + goto disable_clk; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto disable_clk; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1542,17 +1555,30 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto disable_clk; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto disable_clk; + + return 0; + +disable_clk: + clk_disable_unprepare(sgtl5000->mclk); return ret; } static int sgtl5000_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + struct sgtl5000_priv *sgtl5000; + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + snd_soc_unregister_codec(&client->dev); + clk_disable_unprepare(sgtl5000->mclk); return 0; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index a60aaa0..823151b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,20 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - /* assuming clock enabled by default */ - data->codec_clk = NULL; - ret = of_property_read_u32(codec_np, "clock-frequency", - &data->clk_frequency); - if (ret) { - dev_err(&codec_dev->dev, - "clock-frequency missing or invalid\n"); - goto fail; - } - } else { - data->clk_frequency = clk_get_rate(data->codec_clk); - clk_prepare_enable(data->codec_clk); - } + if (IS_ERR(data->codec_clk)) + goto fail; + + data->clk_frequency = clk_get_rate(data->codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -157,10 +147,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -170,7 +160,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } platform_set_drvdata(pdev, data); @@ -179,8 +169,6 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; -clk_fail: - clk_put(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); @@ -194,10 +182,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - if (data->codec_clk) { - clk_disable_unprepare(data->codec_clk); - clk_put(data->codec_clk); - } snd_soc_unregister_card(&data->card); return 0; -- 1.8.1.2

3 6

[alsa-devel] [PATCH] hdspm - remove unneeded semicolon
by Fengguang Wu 15 Jul '13

15 Jul '13

sound/pci/rme9652/hdspm.c:1110:2-3: Unneeded semicolon Generated by: coccinelle/misc/semicolon.cocci Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- cocci-output-13040-790f4c-hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1107,7 +1107,7 @@ static int hdspm_rate_multiplier(struct else if (hdspm->control_register & HDSPM_DoubleSpeed) return rate * 2; - }; + } return rate; }

2 1

[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 15 Jul '13

15 Jul '13

The following changes since commit ad81f0545ef01ea651886dddac4bef6cec930092: Linux 3.11-rc1 (2013-07-14 15:18:27 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc1 for you to fetch changes up to 4824b69a928b6609cfb1cf232ef32d4a8125a371: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-07-15 11:13:58 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 The biggest change here is the OMAP change, these are larger than I'd have liked but make the driver actually usable - during the merge window OMAP removed support for non-DT OMAP4 boards but in doing so removed the method of accessing DMA channels used by the ASoC drivers rendering them unusuable. Otherwise nothing exciting, the symmetric rates change for WM8978 is a fix for the information we expose to userspace. ---------------------------------------------------------------- Fabio Estevam (2): ASoC: sglt5000: Fix the default value of CHIP_SSS_CTRL ASoC: sglt5000: Fix SGTL5000_PLL_FRAC_DIV_MASK Kuninori Morimoto (1): ASoC: wm8978: enable symmetric rates Mark Brown (6): ASoC: wm8994: Remove overly noisy debug logging Merge remote-tracking branch 'asoc/fix/omap' into asoc-linus Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8978' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Padmavathi Venna (1): ASoC: Samsung: Set RFS and BFS in slave mode Peter Ujfalusi (4): ASoC: omap-pcm: Request the DMA channel differently when DT is involved ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/wm8978.c | 1 + sound/soc/codecs/wm8994.c | 4 ---- sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 11 +---------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- sound/soc/samsung/i2s.c | 8 ++++---- 9 files changed, 46 insertions(+), 54 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: ASoC: au1x: fix ac97c module build
by Manuel Lauss 15 Jul '13

15 Jul '13

The recent conversion to module_platform_driver left a stray ampersand in there preventing a build. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Against 3.11-rc1 sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ARM: EXYNOS: remove references to non-DT SoCs
by Paul Bolle 15 Jul '13

15 Jul '13

Commit 383ffda2fa ("ARM: EXYNOS: no more support non-DT for EXYNOS SoCs") removed the Kconfig symbols MACH_SMDKV310 and MACH_SMDKC210. Now remove the references to both of these symbols from the Kconfig entry for SND_SOC_SAMSUNG_SMDK_WM9713. They're not needed anymore. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9855dfc..3d76659 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -136,7 +136,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.1.4

2 3

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot)
by Peter Ujfalusi 15 Jul '13

15 Jul '13

Just recently OMAP4 legacy boot support has been removed. No reason to keep the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5 can boot in this mode. The platform_data header file will be removed at later point to avoid build issues. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- include/linux/platform_data/omap-abe-twl6040.h | 2 + sound/soc/omap/omap-abe-twl6040.c | 133 ++++++++----------------- 2 files changed, 43 insertions(+), 92 deletions(-) diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h index 5d298ac..065abbe 100644 --- a/include/linux/platform_data/omap-abe-twl6040.h +++ b/include/linux/platform_data/omap-abe-twl6040.h @@ -21,6 +21,8 @@ * 02110-1301 USA */ +/* TODO: This file can be removed in 3.12 since it is no longer needed */ + #ifndef _OMAP_ABE_TWL6040_H_ #define _OMAP_ABE_TWL6040_H_ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7..ebb1390 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/mfd/twl6040.h> -#include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> #include <linux/of.h> @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); -- 1.8.2.1

2 4

[alsa-devel] ASoC: codec: SPDIF only works with Tx/Rx seperate codec drivers?
by Nicolin Chen 15 Jul '13

15 Jul '13

Hi Mark, I've a little question about SPDIF codec drivers. I found there're spdif_receiver.c and spdif_transmitter.c, two files, in sound/soc/codecs directory. So it looks like SPDIF module would work separately as one IN device and one OUT device, for example: dts/spear1340.dtsi: spdif-in@d0100000 { dts/spear1340.dtsi: compatible = "st,spdif-in"; dts/spear1340.dtsi: spdif-out@d0000000 { dts/spear1340.dtsi: compatible = "st,spdif-out"; But the question is what if a SPDIF module have both Tx and Rx function in one simple CPU DAI. Is that okay to add a new codec driver for this situation? Actually I'm gonna prepare some patches to add Freescale SPDIF drivers. But this needs to create a brand new dummy SPDIF codec driver, which's some kinda same as spdif_receiver/transmitter.c but quite different in the supports of SAMPLE_RATE/FORMAT and IN/OUT-in-one integration. So I just want to make sure if this'll be okay. Thank you. And looking forward to your reply. Nicolin Chen

3 5

[alsa-devel] Alsa USB Audio regression
by Ulrich Günther 15 Jul '13

15 Jul '13

Hello, i encounterd a possible alsa bug with the "ID 0d8c:0006 C-Media Electronics, Inc. Storm HP-USB500 5.1 Headset". After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't work at all, Ubuntu 11.10 and newer have the bug), the headset only plays sound at maximum volume level if the volume is turned down just a little no sound is being played. I have filled a Ubuntu bug for 12.04 LTS at: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . I was advised to contact this mailing list. AlsaInfo for Ubuntu 12.04 LTS with the bug: http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 . AlsaInfo for Ubuntu 10.04 LTS without the bug: http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc . Thank You for your help.

2 1

Re: [alsa-devel] [PATCH 00/50] USB: cleanup spin_lock in URB->complete()
by Andy Walls 14 Jul '13

14 Jul '13

On Thu, 2013-07-11 at 17:05 +0800, Ming Lei wrote: > Hi, > > As we are going to run URB->complete() in tasklet context[1][2], Hi, Please pardon my naivete, but why was it decided to use tasklets to defer work, as opposed to some other deferred work mechanism? It seems to me that getting rid of tasklets has been an objective for years: http://lwn.net/Articles/239633/ http://lwn.net/Articles/520076/ http://lwn.net/Articles/240054/ Regards, Andy > and > hard interrupt may be enabled when running URB completion handler[3], > so we might need to disable interrupt when acquiring one lock in > the completion handler for the below reasons: > > - URB->complete() holds a subsystem wide lock which may be acquired > in another hard irq context, and the subsystem wide lock is acquired > by spin_lock()/read_lock()/write_lock() in complete() > > - URB->complete() holds a private lock with spin_lock()/read_lock()/write_lock() > but driver may export APIs to make other drivers acquire the same private > lock in its interrupt handler. > > For the sake of safety and making the change simple, this patch set > converts all spin_lock()/read_lock()/write_lock() in completion handler > path into their irqsave version mechanically. > > But if you are sure the above two cases do not happen in your driver, > please let me know and I can drop the unnecessary change. > > Also if you find some conversions are missed, also please let me know so > that I can add it in the next round. > > > [1], http://marc.info/?l=linux-usb&m=137286322526312&w=2 > [2], http://marc.info/?l=linux-usb&m=137286326726326&w=2 > [3], http://marc.info/?l=linux-usb&m=137286330626363&w=2 > [snip] > > > Thanks, > -- > Ming Lei

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: sglt5000: Provide the reg_stride field
by Fabio Estevam 12 Jul '13

12 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 has 16-bit registers, and only even numbers are valid for its registers addresses. Let regmap knows about this feature by specifying the 'reg_stride' field, so that it can access only the valid registers. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the description in the commit log sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7c99f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- 1.8.1.2

2 7

[alsa-devel] [PATCH] ASoC: fsl: Disable SSI in trigger() if RE/TE are both cleared
by Nicolin Chen 12 Jul '13

12 Jul '13

The code enabled SSIEN when triggered by SNDRV_PCM_TRIGGER_START, so move the disable code to SNDRV_PCM_TRIGGER_STOP for symmetric. This also allows us to use the SSI driver more flexible so that it can support some use cases like "aplay S16_LE.wav S24_LE.wav" which would call the driver in sequence like: startup()->hw_params(S16_LE)->trigger(START)->tirgger(STOP)-> hw_params(S24_LE)->trigger(START)->tirgger(STOP)->shutdown() If we disable SSIEN in shutdown(), the second hw_params() would bypass the sample bits setting while using symmetric_rate. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6ab341 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -510,6 +510,9 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); + + if ((read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); break; default: @@ -534,15 +537,6 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; - - /* - * If this is the last active substream, disable the SSI. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - } } static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) -- 1.7.1

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[alsa-devel] [PATCH RFC] ALSA: usb-audio: add front jack channel selector for EMU0204
by Vasily Khoruzhick 12 Jul '13

12 Jul '13

Add support for front jack channel selector which is present on EMU0204. It allows to get 4 channels out of this soundcard. Tested-by: Yury Bushmelev <jay(a)jay-tech.ru> Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- Control request looks like generic UAC_SET_CUR request, but with a bit weird values, valid values are { 0x01, 0x01} and { 0x01, 0x02 }, and there's no such control type ATM. I'm not well familiar with UAC so any suggestins how to get rid of custom control req are welcome! sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 78 insertions(+) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df..4a5f001 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -23,6 +23,7 @@ struct usb_mixer_interface { u8 audigy2nx_leds[3]; u8 xonar_u1_status; + u8 emu0204_ch_switch; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..dc9a397 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -430,6 +430,76 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } +/* EMU0204 */ +#define snd_emu0204_ch_switch_info snd_ctl_boolean_mono_info + +static int snd_emu0204_ch_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = mixer->emu0204_ch_switch; + return 0; +} + +static int snd_emu0204_ch_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int value = ucontrol->value.integer.value[0]; + int err, changed; + unsigned char buf[2]; + + if (value > 1) + return -EINVAL; + + buf[0] = 0x01; + buf[1] = value ? 0x02 : 0x01; + + changed = value != mixer->emu0204_ch_switch; + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) { + err = -ENODEV; + goto out; + } + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x1, + 0x21, + 0x0400, 0x0e00, buf, 2); + out: + up_read(&mixer->chip->shutdown_rwsem); + if (err < 0) + return err; + mixer->emu0204_ch_switch = value; + return changed; +} + + +static struct snd_kcontrol_new snd_emu0204_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Route channels 3/4 to front jack", + .info = snd_emu0204_ch_switch_info, + .get = snd_emu0204_ch_switch_get, + .put = snd_emu0204_ch_switch_put, + .private_value = 0, + }, +}; + +static int snd_emu0204_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i = 0; i < ARRAY_SIZE(snd_emu0204_controls); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_emu0204_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->emu0204_ch_switch = 0; /* Default is 1/2 */ + + return 0; +} /* ASUS Xonar U1 / U3 controls */ static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, @@ -1337,6 +1407,13 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) snd_audigy2nx_proc_read); break; + /* EMU0204 */ + case USB_ID(0x041e, 0x3f19): + err = snd_emu0204_controls_create(mixer); + if (err < 0) + break; + break; + case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); -- 1.8.3.2

2 3

[alsa-devel] Question about WM8904 regcache
by Bo Shen 12 Jul '13

12 Jul '13

Hi Mark, I am sorry to disturb you. On our EK board, it use wm8904 codec. When I enable the audio, and use alsa utils aplay to test, I meet the following issue. The audio can not playback, it show the following error. ---8>--- #aplay 44.1k_16bit_stereo.wav Playing WAVE '44.1k_16bit_stereo.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:1737: write error: Input/output error ---<8--- After check the register, they are not be configured correctly. When read from debug file system, found the cache_dirty always Y before playback and also failed playback. I try to add the following two line in wm8904_set_bias_level into "case SND_SOC_BIAS_ON", it doesn't work. ---8>--- @@ -1832,9 +1834,12 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + regcache_cache_only(wm8904->regmap, false); + regcache_sync(wm8904->regmap); break; ---<8--- However, If I comment the regcache_cache_only(wm8904->regmap, true) as following, then it play OK. ---8>--- @@ -2232,7 +2239,7 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, WM8904_POBCTRL, 0); /* Can leave the device powered off until we need it */ - regcache_cache_only(wm8904->regmap, true); + // regcache_cache_only(wm8904->regmap, true); ---<8--- So, any clue for this issue? Thanks. Best Regards, Bo Shen

2 3

[alsa-devel] [PATCH] ASoC: wm8978: enable symmetric rates
by Kuninori Morimoto 12 Jul '13

12 Jul '13

wm8978 needs .symmetric_rates = 1. The playback/capture will be strange without this patch when it used asymmetric rate in same time Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8978.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 029f31c..d8fc531 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .formats = WM8978_FORMATS, }, .ops = &wm8978_dai_ops, + .symmetric_rates = 1, }; static int wm8978_suspend(struct snd_soc_codec *codec) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/5] ASoC: add simple-card DT support
by Kuninori Morimoto 12 Jul '13

12 Jul '13

Hi Mark, Simon These patches add DT support on simple-card driver. #1 patch adds new snd_soc_of_parse_daifmt() function on soc-core.c and #5 patch will use it. #2 - #5 patches are based on mark/topic/fsi branch >> Simon #2, #3 patches modify sh-arm platform settings. please check it Kuninori Morimoto (5): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: simple-card: add asoc_simple_dai_set for initializing ASoC: simple-card: remove pointless struct asoc_simple_dai_init_info ASoC: simple-card: add Device Tree support .../devicetree/bindings/sound/simple-card.txt | 53 ++++++++ arch/arm/mach-shmobile/board-ap4evb.c | 19 +-- arch/arm/mach-shmobile/board-armadillo800eva.c | 18 +-- arch/arm/mach-shmobile/board-kzm9g.c | 12 +- arch/arm/mach-shmobile/board-mackerel.c | 19 +-- arch/sh/boards/mach-ecovec24/setup.c | 11 +- arch/sh/boards/mach-se/7724/setup.c | 13 +- include/sound/simple_card.h | 11 +- include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 132 ++++++++++++++++---- sound/soc/soc-core.c | 115 +++++++++++++++++ 11 files changed, 307 insertions(+), 98 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v2 0/4] ASoC: Regression fix for OMAP in DT boot (3.11)
by Peter Ujfalusi 11 Jul '13

11 Jul '13

Hi, Changes since v1: - Drop the change in dmaengine_pcm.h - Use .filter_data to store the DMA channel name - omap-pcm: checks if the DAI is probed via DT and uses decides which method to use to request the DMA channel - Dropped the cleanup patch for omap-dmic from the series I will send another series with cleanups later I just noticed in mainline kernel (pre 3.11-rc1) that OMAP4 audio is not probing anymore. This is because OMAP4 also moved to DT only mode (3.10 DT boot was fine). The issue underneath is that we can not use platform_get_resource_byname() for DMA resources when booted with devicetree. In case of DT boot we can use the DMA name to request the channel. Due to technical issues OMAP can not be moved to generic dmaengine pcm as other SoCs did. Mark: can you queue this series for 3.11? Thank you. Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-pcm: Request the DMA channel differently when DT is involved ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 11 +---------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- 4 files changed, 39 insertions(+), 44 deletions(-) -- 1.8.2.1

3 8

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Fix a trivial typo in audss documentation
by Sachin Kamat 11 Jul '13

11 Jul '13

Fixed a trivial typo. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- .../devicetree/bindings/clock/clk-exynos-audss.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt index a120180..75e2e19 100644 --- a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt +++ b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt @@ -2,7 +2,7 @@ The Samsung Audio Subsystem clock controller generates and supplies clocks to Audio Subsystem block available in the S5PV210 and Exynos SoCs. The clock -binding described here is applicable to all SoC's in Exynos family. +binding described here is applicable to all SoCs in Exynos family. Required Properties: -- 1.7.9.5

2 2

Re: [alsa-devel] [PATCH 45/50] sound: usb: usx2y: spin_lock in complete() cleanup
by Takashi Iwai 11 Jul '13

11 Jul '13

At Thu, 11 Jul 2013 22:13:35 +0800, Ming Lei wrote: > > On Thu, Jul 11, 2013 at 9:50 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 11 Jul 2013 17:08:30 +0400, > > Sergei Shtylyov wrote: > >> > >> On 11-07-2013 13:06, Ming Lei wrote: > >> > >> > Complete() will be run with interrupt enabled, so change to > >> > spin_lock_irqsave(). > >> > >> Changelog doesn't match the patch. > > > > Yep, but moreover... > > > >> > Cc: Jaroslav Kysela <perex(a)perex.cz> > >> > Cc: Takashi Iwai <tiwai(a)suse.de> > >> > Cc: alsa-devel(a)alsa-project.org > >> > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > >> > --- > >> > sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ > >> > 1 file changed, 4 insertions(+) > >> > >> > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > >> > index 4967fe9..e2ee893 100644 > >> > --- a/sound/usb/usx2y/usbusx2yaudio.c > >> > +++ b/sound/usb/usx2y/usbusx2yaudio.c > >> > @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) > >> > struct snd_usX2Y_substream *subs = usX2Y->subs[s]; > >> > if (subs) { > >> > if (atomic_read(&subs->state) >= state_PRERUNNING) { > >> > + unsigned long flags; > >> > + > >> > + local_irq_save(flags); > >> > snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); > >> > + local_irq_restore(flags); > >> > } > > > > ... actually this snd_pcm_stop() call should have been covered by > > snd_pcm_stream_lock(). Maybe it'd be enough to have a single patch > > together with the change, i.e. wrapping with > > snd_pcm_stream_lock_irqsave(). > > Please use snd_pcm_stream_lock_irqsave() so that I can avoid sending > out similar patch later, :-) > > > > > I'll prepare the patch for 3.11 independently from your patch series, > > so please drop this one. > > OK, thanks for dealing with that. > > > > > > > BTW, the word "cleanup" in the subject is inappropriate. This is > > rather a fix together with the core change. > > It is a cleanup since the patchset only addresses lock problem which > is caused by the tasklet conversion. Well, the conversion to irqsave() is needed for the future drop of irq_save() in the caller, right? Then this isn't a cleanup but a preparation for movement ahead. > > And, I wonder whether we can take a safer approach. When the caller > > condition changed, we often introduced either a different ops > > (e.g. ioctl case) or a flag for the new condition, at least during the > > transition period. > > Interrupt is't enabled until all current drivers are cleaned up, so it is really > safe, please see patch [2]. OK. > > Last but not least, is a conversion to tasklet really preferred? > > tasklet is rather an obsoleted infrastructure nowadays, and people > > don't recommend to use it any longer, AFAIK. > > We discussed the problem in the below link previously[1], Steven > and Thomas suggested to use threaded irq handler, but which > may degrade USB mass storage performance, so we have to > take tasklet now until we rewrite transport part of USB mass storage > driver. > > Also the conversion[2] has avoided the tasklet spin lock problem > already. OK, good to know. thanks, Takashi > [1], http://marc.info/?t=137079119200001&r=1&w=2 > [2], http://marc.info/?l=linux-usb&m=137286326726326&w=2 > > Thanks, > -- > Ming Lei >

1 0

Re: [alsa-devel] [PATCH 44/50] sound: usb: caiaq: spin_lock in complete() cleanup
by Daniel Mack 11 Jul '13

11 Jul '13

On 11.07.2013 11:06, Ming Lei wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Daniel Mack <zonque(a)gmail.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> Sound right to me, thanks. Acked-by: Daniel Mack <zonque(a)gmail.com> > --- > sound/usb/caiaq/audio.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 7103b09..e5675ab 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -672,10 +672,11 @@ static void read_completed(struct urb *urb) > offset += len; > > if (len > 0) { > - spin_lock(&cdev->spinlock); > + unsigned long flags; > + spin_lock_irqsave(&cdev->spinlock, flags); > fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); > read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); > - spin_unlock(&cdev->spinlock); > + spin_unlock_irqrestore(&cdev->spinlock, flags); > check_for_elapsed_periods(cdev, cdev->sub_playback); > check_for_elapsed_periods(cdev, cdev->sub_capture); > send_it = 1; >

1 0

Re: [alsa-devel] [PATCH 45/50] sound: usb: usx2y: spin_lock in complete() cleanup
by Takashi Iwai 11 Jul '13

11 Jul '13

At Thu, 11 Jul 2013 17:08:30 +0400, Sergei Shtylyov wrote: > > On 11-07-2013 13:06, Ming Lei wrote: > > > Complete() will be run with interrupt enabled, so change to > > spin_lock_irqsave(). > > Changelog doesn't match the patch. Yep, but moreover... > > Cc: Jaroslav Kysela <perex(a)perex.cz> > > Cc: Takashi Iwai <tiwai(a)suse.de> > > Cc: alsa-devel(a)alsa-project.org > > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > > --- > > sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ > > 1 file changed, 4 insertions(+) > > > diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c > > index 4967fe9..e2ee893 100644 > > --- a/sound/usb/usx2y/usbusx2yaudio.c > > +++ b/sound/usb/usx2y/usbusx2yaudio.c > > @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) > > struct snd_usX2Y_substream *subs = usX2Y->subs[s]; > > if (subs) { > > if (atomic_read(&subs->state) >= state_PRERUNNING) { > > + unsigned long flags; > > + > > + local_irq_save(flags); > > snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); > > + local_irq_restore(flags); > > } ... actually this snd_pcm_stop() call should have been covered by snd_pcm_stream_lock(). Maybe it'd be enough to have a single patch together with the change, i.e. wrapping with snd_pcm_stream_lock_irqsave(). I'll prepare the patch for 3.11 independently from your patch series, so please drop this one. BTW, the word "cleanup" in the subject is inappropriate. This is rather a fix together with the core change. And, I wonder whether we can take a safer approach. When the caller condition changed, we often introduced either a different ops (e.g. ioctl case) or a flag for the new condition, at least during the transition period. Last but not least, is a conversion to tasklet really preferred? tasklet is rather an obsoleted infrastructure nowadays, and people don't recommend to use it any longer, AFAIK. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH 36/50] media: usb: em28xx: spin_lock in complete() cleanup
by Devin Heitmueller 11 Jul '13

11 Jul '13

On Thu, Jul 11, 2013 at 5:05 AM, Ming Lei <ming.lei(a)canonical.com> wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: linux-media(a)vger.kernel.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > drivers/media/usb/em28xx/em28xx-core.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/drivers/media/usb/em28xx/em28xx-core.c b/drivers/media/usb/em28xx/em28xx-core.c > index fc157af..0d698f9 100644 > --- a/drivers/media/usb/em28xx/em28xx-core.c > +++ b/drivers/media/usb/em28xx/em28xx-core.c > @@ -941,6 +941,7 @@ static void em28xx_irq_callback(struct urb *urb) > { > struct em28xx *dev = urb->context; > int i; > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -956,9 +957,9 @@ static void em28xx_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->slock); > + spin_lock_irqsave(&dev->slock, flags); > dev->usb_ctl.urb_data_copy(dev, urb); > - spin_unlock(&dev->slock); > + spin_unlock_irqrestore(&dev->slock, flags); > > /* Reset urb buffers */ > for (i = 0; i < urb->number_of_packets; i++) { > -- > 1.7.9.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html I actually stumbled across this a couple of weeks ago, and have had an identical patch running in a local dev tree. Reviewed-by: Devin Heitmueller <dheitmueller(a)kernellabs.com> Tested-by: Devin Heitmueller <dheitmueller(a)kernellabs.com> -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

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[alsa-devel] [PATCH 0/6] ASoC: Regression fix for OMAP in DT boot (3.11)
by Peter Ujfalusi 11 Jul '13

11 Jul '13

Hi, I just noticed in mainline kernel (pre 3.11-rc1) that OMAP4 audio is not probing anymore. This is because OMAP4 also moved to DT only mode (3.10 DT boot was fine). The issue underneath is that we can not use platform_get_resource_byname() for DMA resources when booted with devicetree. In case of DT boot we can use the DMA name to request the channel. Due to technical issues OMAP can not be moved to generic dmaengine pcm as other SoCs did. Mark: can you queue this series for 3.11? Thank you. Regards, Peter --- Peter Ujfalusi (6): ASoC: dmaengine_pcm: Add dma_name to snd_dmaengine_dai_dma_data ASoC: omap-pcm: Request the DMA channel by name when dma_name is set for the dai ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA ASoC: omap-mcbsp: Use different method for DMA request when booted with DT ASoC: omap-dmic: Cleanup the probe function include/sound/dmaengine_pcm.h | 1 + sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- sound/soc/omap/omap-dmic.c | 35 ++++++++++------------------------- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- sound/soc/omap/omap-pcm.c | 16 +++++++++++++--- 5 files changed, 48 insertions(+), 59 deletions(-) -- 1.8.2.1

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[alsa-devel] Problem with MX27 ASoC audio
by Randolph Chung 11 Jul '13

11 Jul '13

I have a target that uses iMX27(L) + tlv320aic3x. The kernel tree is HEAD of linux-arm from about a week ago. Simple playback and recording (using aplay and arecord) work fine. However, I've run into a strange problem when both playback and capture streams are opened at the same time. Consider the attached test program. The capture device is setup with a period time of 10ms, and the program reads samples in a loop. The expected result is that I get a sample every 10ms. This works fine if the playback device is *not* opened (more specifically, if the hw_params are not set). However if the playback device is opened (#if 1 code) then I get samples every 20ms, which seems very wrong. I have tried using both DMA and FIQ ssi-pcm interfaces and both give the same strange result. With the FIQ driver, if I instrument the timer callback, the timer triggers every 10ms but there is only enough data every 20ms. The codec is being driven with a 13MHz MCLK (output from MX27) and the BCLK and WCLK come from the codec. They seem to have the correct rates. Any ideas? Can anybody else with MX27 hardware verify if my test program also fails on their target? thanks, randolph /* Test program */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <assert.h> #include <sys/time.h> int main(int argc, char **argv) { int err = -1; char buf[4096]; unsigned long period_size; snd_pcm_t *in_handle = NULL, *out_handle; snd_pcm_hw_params_t *hwparams; if (snd_pcm_open(&in_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) < 0) { printf("Cannot open capture device\n"); return -1; } snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(in_handle, hwparams); snd_pcm_hw_params_set_access(in_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(in_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(in_handle, hwparams, 1); snd_pcm_hw_params_set_rate(in_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(in_handle, hwparams, 10000, 0); snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0); snd_pcm_hw_params(in_handle, hwparams); snd_pcm_hw_params_free(hwparams); if (snd_pcm_open(&out_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) { printf("Cannot open playback device\n"); return -1; } #if 1 snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(out_handle, hwparams); snd_pcm_hw_params_set_access(out_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(out_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(out_handle, hwparams, 1); snd_pcm_hw_params_set_rate(out_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(out_handle, hwparams, 10000, 0); snd_pcm_hw_params(out_handle, hwparams); snd_pcm_hw_params_free(hwparams); #endif printf("period size = %u\n", period_size); printf("Config done, start loop\n"); while (1) { struct timeval tv; int ret = snd_pcm_readi(in_handle, buf, period_size); if (ret < 0) snd_pcm_prepare(in_handle); gettimeofday(&tv, NULL); printf("[%u.%06u] snd_pcm_readi() return %d\n", tv.tv_sec, tv.tv_usec, ret); } return 0; }

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[alsa-devel] CA0110 analog mic capture flatline
by Ki'Sak 11 Jul '13

11 Jul '13

Good day alsa-devel, I have a "Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG" which I have been unable to capture any content from the analog mic jack. I am a Gentoo Linux user with the 3.8.13 kernel / alsa 1.0.26. This issue has been filed with the wrong devs over at ubuntu's bugtracker and opensuse's forum. Also, I found a report that analog capture was working on this card as recently as September 2009. Thank you for your time, Ki'Sak Note: I have already tried to report this issue, this is a resend without urls due to 2 weeks without a response or it reaching the mailing list.

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[alsa-devel] [PATCH 1/4] ASoC: pcm3008: Remove noisy version print
by Mark Brown 10 Jul '13

10 Jul '13

From: Mark Brown <broonie(a)linaro.org> The version number has never been updated and the printk isn't based on any interaction with the device. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f2a6282..32e5a59 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,8 +28,6 @@ #include "pcm3008.h" -#define PCM3008_VERSION "0.2" - #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -64,8 +62,6 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) struct pcm3008_setup_data *setup = codec->dev->platform_data; int ret = 0; - printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- 1.8.3.2

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[alsa-devel] [PATCH] ALSA: usb-audio: fix regression for fixed stream quirk
by Eldad Zack 10 Jul '13

10 Jul '13

Commit 8f898e92aea2c24c7f379ee265d178f69ebb9c07 removed the redundant reads of bInterfaceProtocol from the descriptors, but introduced a regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT, since fp->protocol is not set in setup process. As a consequence, audio streams would not get initialized, as the following logs show: [ 48.923043] setting usb interface 3:1 [ 48.923056] Creating new capture data endpoint #81 [ 48.923484] 4:3:1: cannot set freq 48000 to ep 0x81 This patch sets fp->protocol in create_fixed_stream_quirk() and resolves the regression. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b01330..1bc45e7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -129,6 +129,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, { struct audioformat *fp; struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; int stream, err; unsigned *rate_table = NULL; @@ -166,6 +167,9 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + altsd = get_iface_desc(alts); + fp->protocol = altsd->bInterfaceProtocol; + if (fp->datainterval == 0) fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) -- 1.8.1.5

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[alsa-devel] [PATCH v2 0/5] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 10 Jul '13

10 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) Tested on at91sam9g35ek. patches applies on next-20130708 [I let the original signed-off from Nicolas and Uwe in place, I don't know if I should replace them by something like "originaly-signed-off-by" since the code has been changed.] [Sorry if it's too late for 3.11] Changes from V1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards Richard Genoud (4): ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: add sound configuration ARM: AT91: DTS: sam9x5ek: enable SSC arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 27 +++++ sound/soc/atmel/Kconfig | 10 ++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 206 +++++++++++++++++++++++++++++++++++ 5 files changed, 248 insertions(+) create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

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[alsa-devel] [PATCH] firewire-speakers: remove not-reused member from structure
by Takashi Sakamoto 09 Jul '13

09 Jul '13

"pcm" member in struct fwspk is used to set pcm operations but is not used again. This commit remove this member and set pcm operations with snd_pcm_set_ops(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/speakers.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 2c63865..fe9e6e2 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -49,7 +49,6 @@ struct fwspk { struct snd_card *card; struct fw_unit *unit; const struct device_info *device_info; - struct snd_pcm_substream *pcm; struct mutex mutex; struct cmp_connection connection; struct amdtp_out_stream stream; @@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk) return err; pcm->private_data = fwspk; strcpy(pcm->name, fwspk->device_info->short_name); - fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - fwspk->pcm->ops = &ops; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); return 0; } -- 1.7.10.4

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Re: [alsa-devel] [PATCH v2 1/5] sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards
by Richard Genoud 09 Jul '13

09 Jul '13

2013/7/8 Mark Brown <broonie(a)kernel.org>: > On Mon, Jul 08, 2013 at 03:29:49PM +0200, Richard Genoud wrote: > >> + * Nicolas Ferre <nicolas.ferre(a)atmel.com> >> + * >> + * Based on sam9g20_wm8731.c by: >> + * Sedji Gaouaou <sedji.gaouaou(a)atmel.com> > > The obvious question here is of course if we can use the same driver for > both of them. I haven't got a g20 to test that, but that's the goal. For now, g20 is still non-DT, so I think it's best to have a DT-only driver like this one for the 9x5 family (9g15, 9g25, 9x25, 9g35, 9x25). When the g20 will move to DT completely, we can drop sam9g20_wm8731.c and adjust sam9x5_wm8731.c (mainly master clock and widgets it seems) By the way, maybe g45 could use that also (and SAMA5 ?) > >> + codec_dai->driver->playback.rates &= SNDRV_PCM_RATE_8000 | >> + SNDRV_PCM_RATE_32000 | >> + SNDRV_PCM_RATE_48000 | >> + SNDRV_PCM_RATE_96000; >> + codec_dai->driver->capture.rates &= SNDRV_PCM_RATE_8000 | >> + SNDRV_PCM_RATE_32000 | >> + SNDRV_PCM_RATE_48000 | >> + SNDRV_PCM_RATE_96000; > > You definitely shouldn't be fiddling with a driver's constant static > data. You want to be using snd_pcm_hw_constraint() APIs to set > additional constraints intead. Ok, I'll change that. Thanks ! -- for me, ck means con kolivas and not calvin klein... does it mean I'm a geek ?

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