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

Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 07 Jul '11

07 Jul '11

At Thu, 7 Jul 2011 16:54:18 +0900, Toshiharu Okada wrote: > > Date: Wed, 06 Jul 2011 13:06:15 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > > > >I just took a quick glance over the code, and wonder whether this kind > >of driver would fit better with ASoC framework. > >Have you considered the implementation on ASoC? > > Thank you for your comment. > > I referred to the following files and implemented this driver. > > http://www.mirrorservice.org/sites/ftp.sourceforge.net/pub/sourceforge/s/pr… > sound/drivers/sta2x11i2s.c > > Therefore, I am not considering about ASoC. > Do I need to fit this driver to an ASoC framework for an upstream? Yes, I guess this would be a better choice in your case. Takashi

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[alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Alex 07 Jul '11

07 Jul '11

Hi, I run a Debian/sid on an Asus P5KPL-SE mainboard, the snd-hda-intel seem to work fine, ever if it run with guess method: Unknown hardware: "HDA-Intel" "Realtek ALC887" "HDA:10ec0887,104383bc,00100202" "0x1043" "0x83bc" Hardware is initialized using a guess method The only annoying thing is about the front headphone, every mixer application show the front headphone item but doesn't have adjustable level and I hear nothing on headphone. Any hint? http://www.alsa-project.org/db/?f=d6bc3005fab4683f2606e48f2bc909258a4ee3e3 Thanks in advance Regards

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[alsa-devel] [PATCH] ASoC: twl6040: Add back support for legacy mode
by Peter Ujfalusi 07 Jul '11

07 Jul '11

The legacy mode has been accidentaly removed by commit: ASoC: twl6040: add all ABE DAIs Add back the twl6040-hifi dai. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Tony, Mark, Liam, this patch fixes the accidental removal of the twl6040-hifi dai from the codec driver. The series, which this patch depends on not yet pulled (I have the branch ready), since I'm still waiting for few acks for it. The MFD series is mostly OK, but the threaded interrupt related followup waits for acks, and also the PLL restructuring series is in waiting state. Is it OK, if I queue up this patch as well for the same pull request, and goes with the rest through l-o (since it is on top of those changes)? I could try to squash it down, if you prefer to the offending commit - I hope it is not breaking the rebase too much... Thank you, Peter sound/soc/codecs/twl6040.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d4b944c..443032b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1464,6 +1464,24 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { + .name = "twl6040-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ .name = "twl6040-ul", .capture = { .stream_name = "Capture", -- 1.7.6

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[alsa-devel] [PATCH v3 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, Changes since v2: - Fix the check in the slaves driver for the validity of the received irq - Mark Brown's ack added to patch 3 Intro mail froim v1: As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Use resource to provide irq number for slaves input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 ++++++--- drivers/mfd/twl6040-core.c | 69 ++++++++++++++++++++++++----------- drivers/mfd/twl6040-irq.c | 16 +-------- include/linux/mfd/twl6040.h | 22 ----------- sound/soc/codecs/twl6040.c | 19 +++++++--- 5 files changed, 74 insertions(+), 69 deletions(-) -- 1.7.6

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Re: [alsa-devel] [RFC] ASoC: Add machine driver for i.MX SGTL500 based boards
by Mark Brown 06 Jul '11

06 Jul '11

On Mon, Jul 04, 2011 at 03:41:55PM -0300, Fabio Estevam wrote: > sgtl5000 1-000a: sgtl5000 revision 17 > sgtl5000 1-000a: asoc: failed to probe CODEC sgtl5000.1-000a: -22 > asoc: failed to instantiate card sgtl5000: -22 Have you looked to see where the error is coming from? > + ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_LRCLK, > + params_rate(params), 0); > + > + if (ret) { > + pr_err("%s: failed setting codec sysclk\n", __func__); > + return ret; > + } Fix this in the CODEC driver, copying this code into the machine drivers is clearly not useful. > +static int __init imx_sgtl5000_init(void) > +{ > + int ret; > + > + imx_sgtl5000_snd_device = platform_device_alloc("soc-audio", -1); > + if (!imx_sgtl5000_snd_device) > + return -ENOMEM; We're not accepting new drivers using soc-audio, you should use a regular platform device and register the card with snd_soc_register_card().

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[alsa-devel] [PATCH] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 06 Jul '11

06 Jul '11

dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- I made this as a cross subsystem patch to maintain bisectability. I suggest to push it either via my tree or the dmaengine tree. drivers/dma/imx-dma.c | 3 ++- drivers/mmc/host/mxcmmc.c | 8 ++++---- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 3 files changed, 8 insertions(+), 7 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/drivers/mmc/host/mxcmmc.c b/drivers/mmc/host/mxcmmc.c index cc20e02..14aa213 100644 --- a/drivers/mmc/host/mxcmmc.c +++ b/drivers/mmc/host/mxcmmc.c @@ -715,13 +715,13 @@ static void mxcmci_set_ios(struct mmc_host *mmc, struct mmc_ios *ios) int burstlen, ret; /* - * use burstlen of 64 in 4 bit mode (--> reg value 0) - * use burstlen of 16 in 1 bit mode (--> reg value 16) + * use burstlen of 64 (16 words) in 4 bit mode (--> reg value 0) + * use burstlen of 16 (4 words) in 1 bit mode (--> reg value 16) */ if (ios->bus_width == MMC_BUS_WIDTH_4) - burstlen = 64; - else burstlen = 16; + else + burstlen = 4; if (mxcmci_use_dma(host) && burstlen != host->burstlen) { host->burstlen = burstlen; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.3 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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[alsa-devel] [PATCH] ASoC: Conditionalize the enable of WM8994 ADC TDM mode
by Mark Brown 06 Jul '11

06 Jul '11

Future devices will not benefit from this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c0d8686..299f6eb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3114,10 +3114,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); - /* Unconditionally enable AIF1 ADC TDM mode; it only affects - * behaviour on idle TDM clock cycles. */ - snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, - WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + /* Unconditionally enable AIF1 ADC TDM mode on chips which can + * use this; it only affects behaviour on idle TDM clock + * cycles. */ + switch (control->type) { + case WM8994: + case WM8958: + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + break; + default: + break; + } wm8994_update_class_w(codec); -- 1.7.5.4

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Re: [alsa-devel] Wrong bitrate from usb microphone
by Takashi Iwai 06 Jul '11

06 Jul '11

At Tue, 05 Jul 2011 15:34:06 +0200, Pali Rohár wrote: > > Hello, > > I have USB webcam with integrated USB microphone. Microphone does not working > correctly because kernel module snd-usb-audio set wrong bitrate. > > Problem appered after this commit: > http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.39.y.git;a=commi… > > When I reverted patch in above commit microphone works properly. > This commit set bitrate to 8000Hz for 041e:4068 but my webcam (propably) > produce 16000Hz. So what to do with my webcam? It needs to revert above patch! > > Here is output from lsusb of my webcam: > Bus 002 Device 003: ID 041e:4068 Creative Technology, Ltd Live! Cam Notebook > [VF0470] > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 1.10 > bDeviceClass 0 (Defined at Interface level) > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 8 > idVendor 0x041e Creative Technology, Ltd > idProduct 0x4068 Live! Cam Notebook [VF0470] > bcdDevice 1.00 > iManufacturer 1 > iProduct 2 > iSerial 0 > bNumConfigurations 1 Could you give the lsusb -v output? Arseniy, also check whether the current code is really OK? thanks, Takashi

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[alsa-devel] [PATCH] Cirrus Logic CS4210-rev-A1 support.
by Tim Howe 06 Jul '11

06 Jul '11

From: Tim Howe <timothyc.howe(a)gmail.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 841 ++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 805 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c7b5ca2..7e09722 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -51,7 +51,7 @@ struct cs_spec { unsigned int cur_adc_format; hda_nid_t dig_in; - const struct hda_bind_ctls *capture_bind[2]; + struct hda_bind_ctls *capture_bind[2]; unsigned int gpio_mask; unsigned int gpio_dir; @@ -61,6 +61,12 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; /* available models */ @@ -114,18 +120,20 @@ enum { static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -231,7 +239,7 @@ static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static const struct hda_pcm_stream cs_pcm_analog_playback = { +static struct hda_pcm_stream cs_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -242,7 +250,7 @@ static const struct hda_pcm_stream cs_pcm_analog_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_analog_capture = { +static struct hda_pcm_stream cs_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -252,7 +260,7 @@ static const struct hda_pcm_stream cs_pcm_analog_capture = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_playback = { +static struct hda_pcm_stream cs_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -264,7 +272,7 @@ static const struct hda_pcm_stream cs_pcm_digital_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_capture = { +static struct hda_pcm_stream cs_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -331,8 +339,8 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t pin = cfg->inputs[idx].pin; - unsigned int val; - if (!is_jack_detectable(codec, pin)) + unsigned int val = snd_hda_query_pin_caps(codec, pin); + if (!(val & AC_PINCAP_PRES_DETECT)) return 0; val = snd_hda_codec_get_pincfg(codec, pin); return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT); @@ -346,10 +354,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { - hda_nid_t pins[2]; + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; unsigned int type; int j, nums; - type = get_wcaps_type(get_wcaps(codec, nid)); + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; if (type != AC_WID_AUD_IN) continue; nums = snd_hda_get_connections(codec, nid, pins, @@ -558,10 +567,10 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, const char *name; int err, index; struct snd_kcontrol *kctl; - static const char * const speakers[] = { + static char *speakers[] = { "Front Speaker", "Surround Speaker", "Bass Speaker" }; - static const char * const line_outs[] = { + static char *line_outs[] = { "Front Line-Out", "Surround Line-Out", "Bass Line-Out" }; @@ -641,7 +650,7 @@ static int build_output(struct hda_codec *codec) /* */ -static const struct snd_kcontrol_new cs_capture_ctls[] = { +static struct snd_kcontrol_new cs_capture_ctls[] = { HDA_BIND_SW("Capture Switch", 0), HDA_BIND_VOL("Capture Volume", 0), }; @@ -709,7 +718,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol, return change_cur_input(codec, idx, 0); } -static const struct snd_kcontrol_new cs_capture_source = { +static struct snd_kcontrol_new cs_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -718,7 +727,7 @@ static const struct snd_kcontrol_new cs_capture_source = { .put = cs_capture_source_put, }; -static const struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, +static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, struct hda_ctl_ops *ops) { struct cs_spec *spec = codec->spec; @@ -821,8 +830,7 @@ static int build_digital_output(struct hda_codec *codec) if (!spec->multiout.dig_out_nid) return 0; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); @@ -847,14 +855,15 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int hp_present; + unsigned int caps, hp_present; hda_nid_t nid; int i; hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - if (!is_jack_detectable(codec, nid)) + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) continue; hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) @@ -923,7 +932,7 @@ static void init_output(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); @@ -982,7 +991,7 @@ static void init_input(struct hda_codec *codec) cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); } -static const struct hda_verb cs_coef_init_verbs[] = { +static struct hda_verb cs_coef_init_verbs[] = { {0x11, AC_VERB_SET_PROC_STATE, 1}, {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, {0x11, AC_VERB_SET_PROC_COEF, @@ -1016,7 +1025,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { * blocks, which will alleviate the issue. */ -static const struct hda_verb cs_errata_init_verbs[] = { +static struct hda_verb cs_errata_init_verbs[] = { {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ @@ -1125,7 +1134,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res) } } -static const struct hda_codec_ops cs_patch_ops = { +static struct hda_codec_ops cs_patch_ops = { .build_controls = cs_build_controls, .build_pcms = cs_build_pcms, .init = cs_init, @@ -1165,7 +1174,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { }; -static const struct snd_pci_quirk cs420x_cfg_tbl[] = { +static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), @@ -1179,7 +1188,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1193,7 +1202,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1207,7 +1216,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1221,22 +1230,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1247,11 +1256,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1278,18 +1289,776 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + Cirrus Logic CS4210 + + 1 DAC => HP(sense) / Speakers, + 1 ADC <= LineIn(sense) / MicIn / DMicIn, + 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 + +/* CS4210 boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +#include <sound/tlv.h> +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + + coef &= ~0x0003; + coef |= (vol & 0x0003); + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + + return 1; +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +/* + CS4210 auto-output redirect + HP/SPK/SPDIF +*/ +static void cs421x_auto_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps, hp_present, spdif_present; + hda_nid_t nid; + int i; + + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + caps = snd_hda_query_pin_caps(codec, nid); + if (caps & AC_PINCAP_PRES_DETECT) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + + hp_present = 0; + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) + continue; + hp_present = snd_hda_jack_detect(codec, nid); + if (hp_present) + break; + } + + /* mute speakers is spdif or hp jack is plugged in */ + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (hp_present | spdif_present) ? 0 : PIN_OUT); + } + + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present & spec->sense_b)? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT:0); + + } + /* Update board GPIOs if neccessary ... */ +} + +/* + CS4210 auto-input redirect + Switch max 3 inputs of a single ADC (nid 3) +*/ +static void cs421x_auto_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + unsigned int present; + + nid = cfg->inputs[spec->automic_idx].pin; + present = snd_hda_jack_detect(codec, nid); + + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); +} + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) + { + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +/* */ +static void init_cs421x_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + /* mute first */ + for (i = 0; i < spec->multiout.num_dacs; i++) + snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (spec->multiout.hp_nid) + snd_hda_codec_write(codec, spec->multiout.hp_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { + if (!spec->multiout.extra_out_nid[i]) + break; + snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + } + + /* set appropriate pin controls */ + for (i = 0; i < cfg->line_outs; i++) + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | HP_EVENT); + spec->hp_detect = 1; + } + } + + /* Speaker */ + for (i = 0; i < cfg->speaker_outs; i++) + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + /* SPDIF is enabled on presence detect */ + if (spec->hp_detect || spec->spdif_detect) + cs421x_auto_output(codec); +} + +static void init_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + unsigned int ctl; + hda_nid_t pin = cfg->inputs[i].pin; + + if (!spec->adc_nid[i]) + continue; + /* set appropriate pin control and mute first */ + ctl = PIN_IN; + if (cfg->inputs[i].type == AUTO_PIN_MIC) { + unsigned int caps = snd_hda_query_pin_caps(codec, pin); + caps >>= AC_PINCAP_VREF_SHIFT; + if (caps & AC_PINCAP_VREF_80) + ctl = PIN_VREF80; + } + + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + snd_hda_codec_write(codec, spec->adc_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(spec->adc_idx[i])); + + if (spec->mic_detect && spec->automic_idx == i) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | MIC_EVENT); + } + } + + if (spec->mic_detect) + cs421x_auto_input(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_cs421x_output(codec); + init_cs421x_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec* spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3) ) + { + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char* name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs421x_auto_output(codec); + break; + + case MIC_EVENT: + cs421x_auto_input(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs ++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", cs421x_models[spec->board_config]); + +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_cirrus[] = { +static struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

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[alsa-devel] [PATCH 0/6] ASoC/MFD: twl6040: PLL handling changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, The following series depends on the v6 of twl6040 MFD series. The aim is to simplify the PLL handling for twl6040: - Move the PLL selection control to the codec driver from machine driver - Sample rate constraint simplification for selected PLL - Avoid multiple, duplicated PLL configuration, which can slow down the audio startup. - Harmonize the PLL ID usage among the drivers, so we avoid redundancy. I have generated this series on top of l-o/devel-cleanup branch + my previous twl6040 MFD/Input/ASoC v6 series, since it depends on the changes introduced there. Is it OK, if Tony will be taking this as well to avoid cross tree merge issues? I can put the series on top of my previous series, and offer Tony a pull link. Regards, Peter --- Peter Ujfalusi (6): ASoC: twl6040: Use neutral name for power mode text/enum ASoC: twl6040: Move PLL selection to codec driver ASoC: twl6040: Simplify sample rate constraint handling ASoC: twl6040: Configure PLL only once MFD: twl6040: Remove enum for PLL tracking ASoC: twl6040: No need to convert the PLL ID drivers/mfd/twl6040-core.c | 26 +++--- include/linux/mfd/twl6040.h | 17 ++--- sound/soc/codecs/twl6040.c | 182 ++++++++++++++++++++++--------------------- sound/soc/codecs/twl6040.h | 1 + sound/soc/omap/sdp4430.c | 47 +---------- 5 files changed, 118 insertions(+), 155 deletions(-) -- 1.7.6

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