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[alsa-devel] [PATCH 2.6.30] ASoC: add additional controls to the CS4270 codec driver
by Timur Tabi 04 Feb '09

04 Feb '09

Update the CS4270 codec driver to allow applications to use the mixer to control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute. Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first initializes the hardware, but these features either don't work or interfere with normal ALSA behavior. However, they can now be re-enabled by an application if desired. Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute bits. The driver previously and erroneously assumed that these bits control only external muting circuitry, but they also control internal muting circuitry, so they should always be used. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/Kconfig | 6 --- sound/soc/codecs/cs4270.c | 93 +++++++++++++++++++------------------------- 2 files changed, 40 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a195303..628a591 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -67,12 +67,6 @@ config SND_SOC_AK4535 config SND_SOC_CS4270 tristate -# Cirrus Logic CS4270 Codec Hardware Mute Support -# Select if you have external muting circuitry attached to your CS4270. -config SND_SOC_CS4270_HWMUTE - bool - depends on SND_SOC_CS4270 - # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2c79a24..cd4a9ee 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -395,17 +395,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Freeze and power-down the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | - CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | - CS4270_PWRCTL_PDN); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Program the mode control register */ + /* Set the sample rate */ reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); @@ -417,7 +407,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } - /* Program the format register */ + /* Set the DAI format */ reg = snd_soc_read(codec, CS4270_FORMAT); reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); @@ -440,42 +430,9 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } - /* Disable auto-mute. This feature appears to be buggy, because in - some situations, auto-mute will not deactivate when it should. */ - - reg = snd_soc_read(codec, CS4270_MUTE); - reg &= ~CS4270_MUTE_AUTO; - ret = snd_soc_write(codec, CS4270_MUTE, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Disable automatic volume control. It's enabled by default, and - * it causes volume change commands to be delayed, sometimes until - * after playback has started. - */ - - reg = cs4270_read_reg_cache(codec, CS4270_TRANS); - reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); - ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Thaw and power-up the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, 0); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - return ret; } -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE /** * cs4270_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI @@ -494,22 +451,23 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) - reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; else - reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); return snd_soc_write(codec, CS4270_MUTE, reg6); } -#else -#define cs4270_mute NULL -#endif /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1), + SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), + SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) }; /* @@ -637,6 +595,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, { struct snd_soc_codec *codec; struct cs4270_private *cs4270; + unsigned int reg; int ret; /* For now, we only support one cs4270 device in the system. See the @@ -702,6 +661,34 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, goto error_free_codec; } + /* Disable auto-mute. This feature appears to be buggy. In some + * situations, auto-mute will not deactivate when it should, so we want + * this feature disabled by default. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = cs4270_i2c_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; + } + + /* Disable automatic volume control. The hardware enables, and it + * causes volume change commands to be delayed, sometimes until after + * playback has started. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; + } + /* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI * structure for each CS4270 device, but the machine driver needs to * have a pointer to the DAI structure, so for now it must be a global -- 1.5.5

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[alsa-devel] [PATCH] ASoC: pxa-ssp: fix SSP port request
by Philipp Zabel 04 Feb '09

04 Feb '09

PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function requested the wrong SSP port. Correcting this unveiled another bug where ssp_init tries to request the already-requested SSP port again. So this patch replaces the ssp_init/exit calls with their internals from mach-pxa/ssp.c, leaving out the redundant ssp_request and the unneeded IRQ request. Effectively, that leaves us with not much more than enabling/disabling the SSP clock. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 15 ++++++++++----- 1 files changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 73cb6b4..f9c6de5 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -21,6 +21,8 @@ #include <linux/clk.h> #include <linux/io.h> +#include <asm/irq.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -221,9 +223,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ); - if (ret < 0) - return ret; + priv->dev.port = cpu_dai->id + 1; + priv->dev.irq = NO_IRQ; + clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } return ret; @@ -238,7 +240,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { ssp_disable(&priv->dev); - ssp_exit(&priv->dev); + clk_disable(priv->dev.ssp->clk); } } @@ -751,7 +753,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id, "SoC audio"); + priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); if (priv->dev.ssp == NULL) { ret = -ENODEV; goto err_priv; -- 1.5.6.5

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[alsa-devel] ASoC: UDA1380/I2S/SSP patches
by Philipp Zabel 04 Feb '09

04 Feb '09

Right now the snd-soc-magician driver in ASoC dev is severely bitrotted. I can make it compile and load again, but until now it only outputs silence (or pops and hisses as I fiddle with the uda1380 registers). For now, I'd like feedback on these patches in preparation for the magician update and switch from pxa2xx-ssp to pxa-ssp. regards Philipp

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[alsa-devel] alsa license question
by Tim Blechmann 04 Feb '09

04 Feb '09

hi all, i am currently working on an alsa driver for the digigram lx6464es interface, based on the sources of the windows driver. since i would like to include some parts of the windows source code, (mainly the header files with register declarations) to the alsa driver, i have a few question: - is it ok, if these files don't match the coding style of the linux kernel? - digigram assured, that i am allowed to include source code to the linux driver. most of the files have the copyright notice: COPYRIGHT (some year) DIGIGRAM. ALL RIGHTS RESERVED. is this statement ok, or do i need to replace the `ALL RIGHTS RESERVED' with a gpl-compatible license note? - is there any repository for alsa driver documentation, provided by the manufacturer? digigram provided the windows driver for the lx6464es, mentioning that this source is `freely accessible to the GNU / ALSA community', so if there is a repository like that, the source package should probably be archived there. thanks, tim -- tim(a)klingt.org http://tim.klingt.org Avoid the world, it's just a lot of dust and drag and means nothing in the end. Jack Kerouac

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[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 04 Feb '09

04 Feb '09

Hello, I was reading the USB Audio Specs from http://www.usb.org/developers/devclass_docs The document talks about Effect Unit Descriptors and few other descriptors.There is no mention of such descriptors in the 'lsusb -v' output for the USB X-Fi 5.1 Does this mean thet the X-Fi,Crystalizer features of the USB X-Fi are implemented purely in software? Here is a link to the lsusb output http://mndar.phpnet.us/usbxfi/files/creative.txt The document "Audio20 final.pdf" also mentions that bDeviceClass, bDeviceSubClass and bDeviceProtocol should have values 0xEF, 0x02, and 0x01 respectively but in the 'lsusb -v' output all three values are 0. Any idea why it is so? Regards Mandar Joshi

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[alsa-devel] [PATCH 1/2] [EXPERIMENTAL] Add support for the latest IndigoIOx and IndigoDJx cards
by Giuliano Pochini 04 Feb '09

04 Feb '09

The following is a very beta patch and is not meant to be merged yet. It has been compile-tested only because I haven't the those cards. Hopefully I'll be able to test it myself soon. As you can see, I reused the existing IndigoIO and IndigoDJ driver because they are the same cards with a different interface. The firmware is different, though. The driver registers the card as a plain -IO or -DJ without the trailing "x" (I will fix it later if someone thinks it's important). diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/echoaudio.h 2009-02-01 21:59:38.000000000 +0100 @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj.c alsa-driver/alsa-kernel/pci/echoaudio/indigodj.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigodj.c 2009-02-01 22:26:02.000000000 +0100 @@ -59,17 +59,21 @@ MODULE_FIRMWARE("ea/loader_dsp.fw"); MODULE_FIRMWARE("ea/indigo_dj_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); #define FW_361_LOADER 0 #define FW_INDIGO_DJ_DSP 1 +#define FW_INDIGO_DJX_DSP 2 static const struct firmware card_fw[] = { {0, "loader_dsp.fw"}, - {0, "indigo_dj_dsp.fw"} + {0, "indigo_dj_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} }; static struct pci_device_id snd_echo_ids[] = { - {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ + {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ */ + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx */ {0,} }; diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-02-01 21:53:55.000000000 +0100 @@ -37,9 +37,11 @@ static int update_vmixer_level(struct ec static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; + u16 subdev; DE_INIT(("init_hw() - Indigo DJ\n")); - if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) + subdev = subdevice_id & 0xfff0; + if (snd_BUG_ON(subdev != INDIGO_DJ && subdev != INDIGO_DJX)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { @@ -50,7 +52,10 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + if (subdev == INDIGO_DJ) + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + else + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio.c alsa-driver/alsa-kernel/pci/echoaudio/indigoio.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigoio.c 2009-02-01 22:26:11.000000000 +0100 @@ -60,17 +60,21 @@ MODULE_FIRMWARE("ea/loader_dsp.fw"); MODULE_FIRMWARE("ea/indigo_io_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); #define FW_361_LOADER 0 #define FW_INDIGO_IO_DSP 1 +#define FW_INDIGO_IOX_DSP 2 static const struct firmware card_fw[] = { {0, "loader_dsp.fw"}, - {0, "indigo_io_dsp.fw"} + {0, "indigo_io_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} }; static struct pci_device_id snd_echo_ids[] = { - {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ + {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO */ + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-02-01 21:59:05.000000000 +0100 @@ -37,9 +37,11 @@ static int update_vmixer_level(struct ec static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; + u16 subdev; DE_INIT(("init_hw() - Indigo IO\n")); - if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) + subdev = subdevice_id & 0xfff0; + if (snd_BUG_ON(subdev != INDIGO_IO && subdev != INDIGO_IOX)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { @@ -50,7 +52,10 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + if (subdev == INDIGO_IO) + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + else + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; -- Giuliano.

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[alsa-devel] asoc: s3c24xx+uda1380 - some questions
by Vasily Khoruzhick 03 Feb '09

03 Feb '09

Hi, I'm trying to develop asoc driver for hp ipaq rx1950 (s3c24xx + uda1380). 1st question: where I can enable/disable uda1380 codec power? I need to control codec power for powersave reasons. I tried to add custom power function call to uda1380_set_bias_level in uda1380.c (pointer to power function is stored in codec->private_data), as it done for uda134x codec, but after enable-disable-enable cycle (for example, after resuming from suspend) driver cannot communicate with codec anymore (i2c bus is always busy), only reloading i2c driver helps. I suspect, power(0) is called in inappropriate moment, when i2c-transfer still is not finished. 2nd question: for some reason, driver does not produce any sound during first aplay invocation, only second and later aplay invocation driver produces sound. What can be the reason of this problem? Regards Vasily

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[alsa-devel] [PATCH] Document tyan model for Realtek ALC262
by Tony Vroon 03 Feb '09

03 Feb '09

As just pointed out to me, the new tyan model for ALC262 was implemented but not documented. This adds the board to the list, using both its marketing name (Thunder n6650W) and its model number (S2915-E). Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 0f5d26b..1aede03 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -56,6 +56,7 @@ ALC262 sony-assamd Sony ASSAMD toshiba-s06 Toshiba S06 toshiba-rx1 Toshiba RX1 + tyan Tyan Thunder n6650W (S2915-E) ultra Samsung Q1 Ultra Vista model lenovo-3000 Lenovo 3000 y410 nec NEC Versa S9100

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[alsa-devel] [PATCH] Add tyan model for Realtek ALC262
by Tony Vroon 03 Feb '09

03 Feb '09

The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a rather odd configuration template. As a result, the white AUX connector can not be used. This rewrites the default config to more accurately reflect the connector layout, colour and function. Unfortunately the black CD_IN connector, which is suspected to be widget 0x1c refuses to switch into input (0x20), instead opting to remain on 0x0. As such, no mixer controls are exposed for it. Autoswitching is implemented between the front headphone output and back line output. Signed-off-by: Tony Vroon <tony(a)linx.net> -- Takashi, this was done on top of 2.6.29-rc3. Please let me know if you'd like it rebased onto anything else instead. Unfortunately I was unable to resolve the recalcitrant widget problem, but at least the default config printed for widget 0x14 makes sense now. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7884a4e..5759195 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -103,6 +103,7 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, + ALC262_TYAN, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -9426,6 +9427,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_tyan_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_tyan_verbs[] = { + /* Headphone automute */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* P11 AUX_IN, white 4-pin connector */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, + + {} +}; + +/* unsolicited event for HP jack sensing */ +static void alc262_tyan_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute line output on ATX panel */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute line output if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_tyan_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_tyan_automute(codec); +} + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -10534,6 +10596,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", + [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; @@ -10574,6 +10637,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), @@ -10792,6 +10856,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_TYAN] = { + .mixers = { alc262_tyan_mixer }, + .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x02, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_tyan_unsol_event, + .init_hook = alc262_tyan_automute, + }, }; static int patch_alc262(struct hda_codec *codec)

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[alsa-devel] [PATCH] ASoC: email - update email addresses.
by Liam Girdwood 03 Feb '09

03 Feb '09

This just updates my email address on some drivers I'd forgotten in a previous patch. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/atmel/atmel_ssc_dai.h | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8990.c | 3 +-- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c5d6790..ff0054b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino(a)endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index a828746..391135f 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino(a)endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 75d3438..d356278 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -3,7 +3,7 @@ * * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 6af1d39..a2e9cc9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -2,8 +2,7 @@ * wm8990.c -- WM8990 ALSA Soc Audio driver * * Copyright 2008 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * lg(a)opensource.wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 -- 1.5.6.3

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