February 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] PATCH: Maxdata Favorit 100XS support
by Michael Schwingen 05 Feb '09

05 Feb '09

Hi, this is my first ALSA patch, so I am not sure if this is the right place & form to post patches? Anyway, here is a patch to enable audio on the Maxdata Favorit 100XS - a compact PC sold by German company Maxdata (probably manufactured by FIC, Transtec Senyo 600/Belinea o.max XS/Lynx Silent PC MP-I *might* be identical, but I have none of those). The Favorit 100XS uses an Intel 945 chipset with a Realtek HD audio codec connected to exactly 2 jacks: one input, and one output jack. Using the standard ALSA driver, output is silent. This patch enables audio input/output on these machines - it is based in part on the acer code in the same file. Signed-off-by: Michael Schwingen <michael(a)schwingen.org> cu Michael

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[alsa-devel] [PATCH 0/3] Add AVR32 ALSA drivers
by Hans-Christian Egtvedt 05 Feb '09

05 Feb '09

From: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> This series adds support for the ABDAC and AC97C peripherals on AVR32 devices. They also need the DW DMAC cyclic DMA API patch to work[1]. The patch set is made against kernel v2.6.29-rc3. A review would be greatly appreciated, checkpatch.pl gives no warnings and usage works as expected on hardware. Adding these drivers to the queue for the upstream kernel should wait until the cyclic DMA API for the DW DMAC driver is also in the queue for upstream. 1: http://avr32linux.org/archives/kernel/2009-February/002088.html http://avr32linux.org/archives/kernel/2009-February/002087.html -- Best regards, Hans-Christian Egtvedt

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[alsa-devel] [PATCH] hda: Added stac378x digital slave out struct
by Matthew Ranostay 05 Feb '09

05 Feb '09

Added the ADATOut nid to a slave digital outs struct to allow output via the DigOut pin. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a7df81e..58a10ed 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -404,6 +404,10 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +static hda_nid_t stac927x_slave_dig_outs[2] = { + 0x1f, 0, +}; + static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -5218,6 +5222,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, -- 1.5.6.3

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[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 05 Feb '09

05 Feb '09

>>Does this mean thet the X-Fi,Crystalizer features of the USB X-Fi are implemented purely in software? I think this might be the case. I started SniffUSB under WindowsXP and tried changing the settings using the audio console software provided by Creative. When changing the Speaker selection (2.1/4.1/5.1), data is sent over usb to the device. However, when I try to change the settings for the X-Fi Crystalizer or EAX or CMSS-3D or Graphic Equalizer nothing gets sent to the device but the audio output does change and I am able to hear the corresponding effects. And as mentioned in my previous post, the lsusb output doesn't show any effect descriptors or feature units If this is in fact true that all the xtreme fidelity(X-Fi) stuff is happening in software alone, my purchase has been a total waste!! Does anybody on this list have a USB X-Fi 5.1 that they can use to verify my observations? BTW, the mailing list seems to be creating separate threads for every post I make even though the subject line is the same. What am I missing? I'm using Yahoo Mail. Regards Mandar Joshi

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[alsa-devel] Finding the buffer fill level
by Bankim Bhavsar 05 Feb '09

05 Feb '09

Hi Jaroslav, I'm using following as the reference to find buffer fill-level of the playback channel * WRITE_TO_READ: the current fill level of the playback buffer - buffer_size - snd_pcm_avail_update() http://article.gmane.org/gmane.linux.alsa.devel/54017 Infact I use snd_pcm_hwsync() before calling snd_pcm_avail_update() ( I know I can use snd_pcm_avail() but thats only available 1.0.18 onwards) I use snd_pcm_set_buffer_size_near() to configure the hw params and save the returned value of bufffer_size to find the buffer fill level later on. On some sound cards ( snd_pcm_avail_update() > buffer_size) which shouldn't be the case! This throws off our internal polling calculations. Information about the Linux hosts and sound cards where we've seen this issue. Ubuntu 8.10 x64 host with kernel version 2.6.27-11-generic Alsa lib version 1.0.17a 1) **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] 2) **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC885 Digital [ALC885 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 Is this a known issue? Let me know if you need more information. Or there is an alternative approach/workaround? Thanks, Bankim.

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[alsa-devel] [PATCH] pcm_oss: AFMT_S24_LE is set twice in return value
by Roel Kluin 04 Feb '09

04 Feb '09

vi sound/core/oss/pcm_oss.c +640 #define AFMT_S24_LE 0x00008000 #define AFMT_S24_BE 0x00010000 ---------------------->8----------------------8<------------------- AFMT_S24_LE is set twice in return value Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e178366..0a1798e 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE | AFMT_S32_LE | AFMT_S32_BE | - AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_LE | AFMT_S24_BE | AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params)

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Re: [alsa-devel] No headphone with an Intel 82801I sound card
by Aurélien Croc 04 Feb '09

04 Feb '09

Is there no news about that problem? If it is just a mixer problem, it may not be so hard to get the correct mixer channels available no? (in particular when each point of the problem work with a specific code). Thank you in advance, Best regards, Aurélien On Monday 02 February 2009, Takashi Iwai wrote: > At Mon, 2 Feb 2009 16:56:07 +0100, > > Aurélien Croc wrote: > > > Hmm, other people reported basically such Dell laptops do work with > > > the headphone but not with the speaker output... > > > > With google i found a lot of guys who have the same problem...I never > > seen someone who doesn't have its internal speaker working.. > > See the recent post by Laurent Pinchart. Which reminds me that the problem still hasn't been fixed :-) Is there anything else you would like me to try ? > Did you try the latest snapshot? > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna >pshot.tar.gz Best regards, Laurent Pinchart _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] sscape: drop redundant fields from soundscape struct (3rd rev)
by Krzysztof Helt 04 Feb '09

04 Feb '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The wss_base is disuised parameter for one function. It is converted to function parameter. The code_type is only set but never read. It is removed. The midi_vol is set only to 0 so it does not work as detection of change in midi volume. It is fixed. The xport variable is alias to the port[dev]. Use the port[dev] directly to increase readability. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The sscape_midi_get() now returns the s->midi_vol value. sound/isa/sscape.c | 44 ++++++++++++++++---------------------------- 1 files changed, 16 insertions(+), 28 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 681e223..33c1258 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -135,8 +135,6 @@ enum card_type { struct soundscape { spinlock_t lock; unsigned io_base; - unsigned wss_base; - int codec_type; int ic_type; enum card_type type; struct resource *io_res; @@ -726,13 +724,7 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, unsigned long flags; spin_lock_irqsave(&s->lock, flags); - set_host_mode_unsafe(s->io_base); - - if (host_write_ctrl_unsafe(s->io_base, CMD_GET_MIDI_VOL, 100)) { - uctl->value.integer.value[0] = host_read_ctrl_unsafe(s->io_base, 100); - } - - set_midi_mode_unsafe(s->io_base); + uctl->value.integer.value[0] = s->midi_vol; spin_unlock_irqrestore(&s->lock, flags); return 0; } @@ -767,6 +759,7 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); + s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; __skip_change: /* @@ -809,12 +802,11 @@ static unsigned __devinit get_irq_config(int irq) * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. */ -static int __devinit detect_sscape(struct soundscape *s) +static int __devinit detect_sscape(struct soundscape *s, long wss_io) { unsigned long flags; unsigned d; int retval = 0; - int codec = s->wss_base; spin_lock_irqsave(&s->lock, flags); @@ -830,13 +822,11 @@ static int __devinit detect_sscape(struct soundscape *s) if ((d & 0x80) != 0) goto _done; - if (d == 0) { - s->codec_type = 1; + if (d == 0) s->ic_type = IC_ODIE; - } else if ((d & 0x60) != 0) { - s->codec_type = 2; + else if ((d & 0x60) != 0) s->ic_type = IC_OPUS; - } else + else goto _done; outb(0xfa, ODIE_ADDR_IO(s->io_base)); @@ -856,10 +846,10 @@ static int __devinit detect_sscape(struct soundscape *s) sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); if (s->type == SSCAPE_VIVO) - codec += 4; + wss_io += 4; /* wait for WSS codec */ for (d = 0; d < 500; d++) { - if ((inb(codec) & 0x80) == 0) + if ((inb(wss_io) & 0x80) == 0) break; spin_unlock_irqrestore(&s->lock, flags); msleep(1); @@ -1057,7 +1047,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) unsigned dma_cfg; unsigned irq_cfg; unsigned mpu_irq_cfg; - unsigned xport; struct resource *io_res; struct resource *wss_res; unsigned long flags; @@ -1077,15 +1066,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } - xport = port[dev]; /* * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - io_res = request_region(xport, 8, "SoundScape"); + io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport); + snd_printk(KERN_ERR + "sscape: can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; @@ -1112,10 +1101,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; - sscape->io_base = xport; - sscape->wss_base = wss_port[dev]; + sscape->io_base = port[dev]; - if (!detect_sscape(sscape)) { + if (!detect_sscape(sscape, wss_port[dev])) { printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); err = -ENODEV; goto _release_dma; @@ -1188,11 +1176,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, xport, mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%x\n", - xport); + "MPU-401 device at 0x%lx\n", + port[dev]); goto _release_dma; } -- 1.5.2.2 ---------------------------------------------------------------------- Obdaruj swoja Walentynke ... lub siebie! Wygraj nagrody! Sprawdz >> http://link.interia.pl/f203a

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[alsa-devel] [PATCH] hda: Add digital beep generator support for Realtek codecs.
by Kusanagi Kouichi 04 Feb '09

04 Feb '09

A digital beep generator can be used via input layer. Signed-off-by: Kusanagi Kouichi <slash(a)ma.neweb.ne.jp> --- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index b9679f0..51bf6a5 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -39,7 +39,7 @@ struct hda_beep { int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) +#define snd_hda_attach_beep_device(...) 0 #define snd_hda_detach_beep_device(...) #endif #endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7884a4e..956b327 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -3160,6 +3161,7 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); + snd_hda_detach_beep_device(codec); codec->spec = NULL; /* to be sure */ } @@ -4328,6 +4330,12 @@ static int patch_alc880(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); @@ -5855,6 +5863,12 @@ static int patch_alc260(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); @@ -7050,6 +7064,12 @@ static int patch_alc882(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); @@ -9024,6 +9044,12 @@ static int patch_alc883(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); @@ -10844,6 +10870,12 @@ static int patch_alc262(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); @@ -11896,6 +11928,12 @@ static int patch_alc268(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); @@ -12730,6 +12768,12 @@ static int patch_alc269(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); @@ -13823,6 +13867,12 @@ static int patch_alc861(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); @@ -14769,6 +14819,12 @@ static int patch_alc861vd(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); @@ -16578,6 +16634,12 @@ static int patch_alc662(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); -- 1.5.6.5

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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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