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[alsa-devel] linux vs c99 fixed width types
by Eliot Blennerhassett 05 Jul '10

05 Jul '10

Hi Takashi, & list it is quite likely that we will be moving our own codebase to use the C99 fixed width types rather than our own e.g. uint32_t vs HW32 etc. Currently HW32 is translated into u32 before I submit patches to you. Should I do the same for the C99 types, or let them pass through unaltered? regards -- Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] Update asihpi to hpi version 4.03.36
by linux＠audioscience.com 05 Jul '10

05 Jul '10

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 05 Jul '10

05 Jul '10

The following changes since commit e827e32efc313e1a061c81007e96fff439ccb802: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2010-06-24 11:11:41 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Eric Bénard (1): ASoC: clean i.MX Kconfig Kuninori Morimoto (1): ASoC: ak4642: Add Digital Playback Volume control Mark Brown (1): Merge remote branch 'takashi/topic/asoc' into for-2.6.36 Vladimir Zapolskiy (4): ASoC: uda134x: replace a macro with a value in platform struct. ASoC: uda134x: fix bias level setup on initialization ASoC: uda134x: add DATA011 register found in codecs family ASoC: uda134x: correct bias level setup for codecs family include/sound/uda134x.h | 12 ++++++++ sound/soc/codecs/ak4642.c | 24 ++++++++++++++-- sound/soc/codecs/uda134x.c | 64 +++++++++++++++++++++++++++++-------------- sound/soc/codecs/uda134x.h | 5 ++- sound/soc/imx/Kconfig | 14 ++++----- 5 files changed, 85 insertions(+), 34 deletions(-)

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[alsa-devel] PC Beeps not working with HDA Intel module
by Shérab 05 Jul '10

05 Jul '10

Dear all, The problem is on a computer with an HDA Intel sound card, Intel G45 DEVCTG chip and appears will all version s of the kernel. Running e.g. beep 440 produces a beep whose audible frequency is very low, definitely not 440 Hz. However, it is worth notiing that grub is able to produce a beep of the right frequency at boot time. Can this problem be fixed ? Should you need any additional bit of information or testing, please ask. Many thanks in advance for your help, Shérab.

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[alsa-devel] Full duplex code sample?
by robert 05 Jul '10

05 Jul '10

Dear All, Is there an example code on how to program a full duplex audio transmission? I did search the internet but it is full of Paul Davis advising to use Jack instead of ALSA. I think I would rather use ALSA for my purposes. Cheers, Robert

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[alsa-devel] [RFC][PATCH] snd-usb: improving support for CM6206
by Adrian Pardini 05 Jul '10

05 Jul '10

Hi all, I'm working in my spare time to get better support for this chip. Right now it works but some features present in the datasheet are missing. This patch gives the option to select the source for the "headphone" jack. I have some concerns with the way I accomplished this and I'd like if someone could kick me in the right direction. In particular with the caching of some register values. Added another field to usb_mixer_interface, maybe it's not the best place. Also, according to the datasheet, in order to read back the register I must issue a request as done in snd_usb_cm106_write_int_reg() and the result will come in the HID input report. For this it doesn't matter but I'm planning on making some controls for spdif and I'll need it. Where should I tap to get that information?. To be done (I need help): * Mapping of streams: Front is ok. Rear_L jack is assigned to "Center", Rear_R jack gets "Speaker Woofer". "Center/Bass" jack is managed with the mixer "Speaker". Traced the circuit and the connections are fine, how do I switch them from ALSA? (nb: not with a ttable, from the driver) To be done (I think I'll somehow get this done on my own): * Input levels are reported with more resolution than the real. Add another case to build_feature_ctl(). Not very important now. * Engaging of the "mute" controls: now I can't mute left or right channel without muting everything. Have to play more with mixer_maps. Kind regards and thanks a lot for your time. Signed off by: Adrian Pardini <adrian.pardini(a)solar.org.ar> --- diff --git a/alsa-kernel/usb/mixer.h b/alsa-kernel/usb/mixer.h index 26c636c..291ba4b 100644 --- a/alsa-kernel/usb/mixer.h +++ b/alsa-kernel/usb/mixer.h @@ -22,6 +22,7 @@ struct usb_mixer_interface { u8 audigy2nx_leds[3]; u8 xonar_u1_status; + u16 cm6206reg2; }; #define MAX_CHANNELS 10 /* max logical channels */ diff --git a/usb/mixer_quirks.c b/usb/mixer_quirks.c index 794cdbd..8adeef0 100644 --- a/usb/mixer_quirks.c +++ b/usb/mixer_quirks.c @@ -360,6 +360,129 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely + * documented in the device's data sheet. + */ +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) +{ + u8 buf[4]; + buf[0] = 0x20; + buf[1] = value & 0xff; + buf[2] = (value >> 8) & 0xff; + buf[3] = reg; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, + USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, + 0, 0, &buf, 4, 1000); +} + +static int snd_cm6206_headphone_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value; + return 0; +} + +static int snd_cm6206_headphone_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_device *dev = mixer->chip->dev; + + u16 reg2 = mixer->cm6206reg2; + u16 idx = kcontrol->private_value = ucontrol->value.integer.value[0]; + + reg2 = (u16)( (((1<<7) | (idx<<5))<<8) | (reg2 & 0x1800) ); + + snd_usb_cm106_write_int_reg(dev, 2, reg2); + mixer->cm6206reg2 = reg2; + return 1; +} + +static int snd_cm6206_headphone_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value & (1<<11); + ucontrol->value.integer.value[1] = kcontrol->private_value & (1<<12); + return 0; +} + +static int snd_cm6206_headphone_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_device *dev = mixer->chip->dev; + + u16 reg2 = mixer->cm6206reg2; + u16 mute; + + mute = ucontrol->value.integer.value[1] ? (1<<12) : 0; + mute |= ucontrol->value.integer.value[0] ? (1<<11) : 0; + kcontrol->private_value = mute; + + reg2 = (u16)(mute | (reg2 & ~0x1800)); + + snd_usb_cm106_write_int_reg(dev, 2, reg2); + mixer->cm6206reg2 = reg2; + return 1; +} + +static int snd_cm6206_headphone_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const names[] = { + "Side", + "Surround", + "Center", + "Front" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, names[uinfo->value.enumerated.item]); + return 0; +} + +#define snd_cm6206_headphone_mute_info snd_ctl_boolean_stereo_info + +static struct snd_kcontrol_new cm6206_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "Headphone Source", + .info = snd_cm6206_headphone_source_info, + .get = snd_cm6206_headphone_source_get, + .put = snd_cm6206_headphone_source_put, + .private_value = 3 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "Headphone", + .info = snd_cm6206_headphone_mute_info, + .get = snd_cm6206_headphone_mute_get, + .put = snd_cm6206_headphone_mute_put, + .private_value = 24 + } +}; + +static int snd_cm6206_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i=0; i < ARRAY_SIZE(cm6206_controls); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&cm6206_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->cm6206reg2 = 0xf800; /* Headphone source=front, no mute. */ + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err; @@ -385,6 +508,11 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; } + if (mixer->chip->usb_id == USB_ID(0x0d8c, 0x0102)) { + err = snd_cm6206_controls_create(mixer); + if (err < 0) + return err; + } return 0; } @@ -415,4 +543,3 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, break; } } - diff --git a/alsa-kernel/usb/mixer_quirks.h b/alsa-kernel/usb/mixer_quirks.h index bdbfab0..bc5d577 100644 --- a/alsa-kernel/usb/mixer_quirks.h +++ b/alsa-kernel/usb/mixer_quirks.h @@ -9,5 +9,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid); +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value); + #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/usb/quirks.c b/usb/quirks.c index bf97ccd..ab57d58 100644 --- a/usb/quirks.c +++ b/usb/quirks.c @@ -376,22 +376,6 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) return 0; } -/* - * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely - * documented in the device's data sheet. - */ -static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) -{ - u8 buf[4]; - buf[0] = 0x20; - buf[1] = value & 0xff; - buf[2] = (value >> 8) & 0xff; - buf[3] = reg; - return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, - USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); -} - static int snd_usb_cm106_boot_quirk(struct usb_device *dev) { /* -- Adrian. http://elesquinazotango.com.ar http://www.noalcodigodescioli.blogspot.com/

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Re: [alsa-devel] [PATCH 2/6] sound/oss: convert to unlocked_ioctl
by Sam Ravnborg 04 Jul '10

04 Jul '10

> > diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c > index c1070e3..3547450 100644 > --- a/sound/oss/au1550_ac97.c > +++ b/sound/oss/au1550_ac97.c > @@ -824,22 +824,26 @@ mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, > return codec->mixer_ioctl(codec, cmd, arg); > } > > -static int > -au1550_ioctl_mixdev(struct inode *inode, struct file *file, > - unsigned int cmd, unsigned long arg) > +static long > +au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) > { > struct au1550_state *s = (struct au1550_state *)file->private_data; > struct ac97_codec *codec = s->codec; > + int ret; > + > + lock_kernel(); > + ret = mixdev_ioctl(codec, cmd, arg); > + unlock_kernel(); > > - return mixdev_ioctl(codec, cmd, arg); > + return ret; > } General comment... In several places you declare a local variable of type "int", but the function return a long. I know that mixdev_ioctl() return an int so nothing is lost but it just looks wrong. > diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c > index 3f3c3f7..b5464fb 100644 > --- a/sound/oss/dmasound/dmasound_core.c > +++ b/sound/oss/dmasound/dmasound_core.c > @@ -337,8 +337,8 @@ static int mixer_release(struct inode *inode, struct file *file) > unlock_kernel(); > return 0; > } > -static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, > - u_long arg) > + > +static long mixer_ioctl(struct file *file, u_int cmd, u_long arg) > { > if (_SIOC_DIR(cmd) & _SIOC_WRITE) > mixer.modify_counter++; > @@ -362,11 +362,22 @@ static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, > return -EINVAL; > } > > +static long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) > +{ > + int ret; > + > + lock_kernel(); > + ret = mixer_ioctl(file, cmd, arg); > + unlock_kernel(); > + > + return ret; > +} Here it looks like a potential but. mixer_ioctl() return a long which is stored in an int and then we return a long. > -static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, > - u_long arg) > +static long sq_ioctl(struct file *file, u_int cmd, u_long arg) > { > int val, result; > u_long fmt; > @@ -1114,18 +1124,29 @@ static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, > return IOCTL_OUT(arg,val); > > default: > - return mixer_ioctl(inode, file, cmd, arg); > + return mixer_ioctl(file, cmd, arg); > } > return -EINVAL; > } > > +static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) > +{ > + int ret; > + > + lock_kernel(); > + ret = sq_ioctl(file, cmd, arg); > + unlock_kernel(); > + > + return ret; > +} ditto: long -> int -> long Sam

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[alsa-devel] ESI DuaFire Vendor Information
by Florian Scholz 04 Jul '10

04 Jul '10

Hi I like to implement a driver for the ESI DuaFire Interface and need the information from ESI. Andreas Preuss from ESI wrote me (translated): thanks you for your message. every information that you need for the development of a firewire audio driver are public. the information are accessable for the alsa-project and maybe the ffado project.

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[alsa-devel] [PATCH 1/3] ASoC: codec: Add WM8987 device id to WM8750 driver
by Maurus Cuelenaere 04 Jul '10

04 Jul '10

The WM8987 codec is register compatible with the WM8750, so just add it to the SPI and I²C device table. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/codecs/wm8750.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9407e19..e2c05e3 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8750_i2c_id[] = { { "wm8750", 0 }, + { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */ { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); @@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_id[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750", + .name = "WM8750 SPI Codec", .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), + .id_table = wm8750_spi_id, }; #endif -- 1.7.0.4

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[alsa-devel] Question about snd_pcm_sw_params_set_start_threshold
by Christian Gmeiner 03 Jul '10

03 Jul '10

Hi all, I am running into some troubles with my linux plugin for the vdr. I want to play audio in sync of video. My current solution is to buffer about 500ms of video on the output device (pts based FIFO). I can play audio with alsa, but sometimes I am not in sync with the video. So I want to set start threshold to 0 to disable automatic playback start. Instead I want to buffer audio and start playback manual, if pts of video is near current scr clock of output device. What are my options to manually start playback? thanks, -- Christian Gmeiner, MSc

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