February 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 000/102 - chunk#5] Add SOC_ENUM_SINGLE_CONST() & co
by Takashi Iwai 21 Feb '14

21 Feb '14

This is the last chunk of patch series, which introduces two new macros, SOC_ENUM_SINGLE_CONST() and SOC_ENUM_DOUBLE_CONST() in order to simplify the codes without ARRAY_SIZE() in the caller side. Takashi

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Re: [alsa-devel] Headphone output problems with HP ProDesk 600 G1 (Realtek ALC221)
by Raymond Yau 20 Feb '14

20 Feb '14

>> I have a HP ProDesk 600 G1 machine which seems to have some strange behavior with the front headphone jack(s) running under Fedora 20 (running 3.13.2 kernel): >> This machine has two front jacks: a headphone jack and a microphone/headphone jack. When the headphones are plugged into the headphone jack, there's usually no change in the sound outputs list in the Sound control panel. The audio (mostly) works if the audio output device is set to Analog Output, but sometimes spuriously changes back to Speakers and the headphone output stops working (I suspect maybe from a spurious jack-sensing event, though I'm not certain). More confusingly, a couple of times I have gotten it into the state (through various fiddling with jacks and playing in the Sound control panel) where plugging into this jack is detected and causes the Headphone output to be selected in the control panel. I haven't been able to figure out consistently what causes this to happen. >> If the headphone is plugged into the microphone/headphone jack, then the control panel consistently switches to the Headphone output and consistently switches back the way it was when it's unplugged. However, there's no actual audio output coming out of that jack. (From some of the HP manuals, it sounds like with the Windows driver, when you plug into this jack, it's supposed to pop up a prompt asking you whether you want to use it as a microphone or headphones.) >> The other strange thing is that intermittently, on the headphone jack you'll start getting a bunch of loud static (clicking noises) and any actual audio playback becomes very faint and distorted. >> I'm guessing perhaps something isn't being initialized/setup properly in the codec, causing this kind of erratic behavior. The alsa-info output is uploaded at: http://www.alsa-project.org/db/?f=594e837b7af07ffbe3473cf807465bc81a587880 >> I tried using hda-analyzer but it fails on this machine with: >> ValueError: wrong proc file format (unknown dig1 bit 'KAE') >> Any suggestions for things to try to debug this? Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 it is a driver bug which assign PCM playback Switch to Line Out Jack try the patch https://bugzilla.kernel.org/attachment.cgi?id=124971 HP ProDesk 600 G1 SFF is an All-In-One pc , should headphone and line out share the volume control if headphone can mute the line out or speaker Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Speaker Only' 'Line Out Speaker' Item0: 'Line Out Speaker' Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC221 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x57 0x57] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM

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Re: [alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by David Henningsson 20 Feb '14

20 Feb '14

On 02/19/2014 03:57 PM, Takashi Sakamoto wrote: > Hi all, > > I have two questions about the usage of this macro. > > Should I use this macro for new drivers which I'm working for?' > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… > > > My drivers support some firewire devices, which is desinged for > studio-use and have many channels. Its combination of channels are not > designed for surround sound like 5.1ch. > > Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. > But there are no drivers which use this macro in current tree, against > my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so > on) should use this macro. > > Are there any reasons? I've never heard of it before. I had a quick look in alsa-lib and couldn't find any difference in handling between CLASS_GENERIC and CLASS_MULTI. It's probably something historic that nobody uses any more. Maybe people who have been along longer knows what it is. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 20 Feb '14

20 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 490 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 507 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..847e079 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,490 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used[SNDRV_CARDS/sizeof(int)]; +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int len) +{ + unsigned int length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!(devices_used[card_index/sizeof(int)*8] & + (1 << card_index % (sizeof(int)*8)))) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used[card_index/sizeof(int)*8] |= + 1 << (card_index%(sizeof(int)*8)); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used[bcd2k->card_index/sizeof(int)*8] &= + ~(1 << (bcd2k->card_index%(sizeof(int)*8))); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

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[alsa-devel] [PATCH v4 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 20 Feb '14

20 Feb '14

Hi, These are the remaining patches from v3 of this version. They add master clock and regulator support. In v4 I added a check for the regulator dependencies and updated the binding documentation for them. I fixed the master clock patch to shutdown the master clock after the PLL and added another patch to shutdown all dividers before the PLL. Regards, Markus Changes in v4: - Fix clock shutdown order - Add regulator dependency checks - Revert the previously added tlv320aic32x4 compatible from the binding documentation Markus Pargmann (4): ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown Revert "ASoC: codec doc, tlv320aic3x: Add tlv320aic32x4 as compatible" .../devicetree/bindings/sound/tlv320aic32x4.txt | 12 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - sound/soc/codecs/tlv320aic32x4.c | 126 ++++++++++++++++++--- 3 files changed, 123 insertions(+), 16 deletions(-) -- 1.8.5.3

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[alsa-devel] Need Help: Repeated Jackhammer noise after several hours of play and heavy cpu usage in snd_pcm_dmix_sync_area
by Adarsh 20 Feb '14

20 Feb '14

Hi, We are facing a issue where if gdb is attached to the our long running sound process and continued, jackhammer sound is played over and over again abusing the speakers. Process details: 1. We are using RW_INTERLEAVED access as described in write_loop method in /test/pcm.c and opening 'default' device. 2. However to disable the underruns, the snd_pcm_sw_params_set_stop_threshold is set to 0x7fffffff which is greater than boundary value. 3.Period size is 1024 and buffer size is 2048. We try to write period_size no of frames using snd_pcm_writei. (I am attaching a sample program which shows this behaviour.) Steps to Reproduce: 1. Left process running for over 12 hours. 2. gdb is attached to the process and continued inside gdb. 3. At this point, the thread went into a tight loop taking >100% cpu and repeatedly playing same sound. It did not recover for a long time. Stack trace shows that 'snd_pcm_dmix_sync_area' in pcm_dmix.c is taking a very long time to return because the 'size' is a large unsinged interger and mix_areas is getting called with 'transfer' value of <=2048. If snd_pcm_sw_params_set_stop_threshold is not set or when application is running for shorter duration, attaching gdb does not show this behaviour. Is it that setting of stop threshold to 0x7fffffff is triggering this and attaching gdb is making the buffer pointers misbehave? If so, what can be done if we do not want PCM to stop? I have tried by setting silence threshold and silence size to let alsa play silence by itself but still observed the issue. Below is the stack trace and PCM dump: (gdb) bt #0 0xb7c44463 in mix_areas_32_smp (size=1360, dst=0xb6855000, src=0x84a7510, sum=0xb6845000, dst_step=4, src_step=4, sum_step=4) at pcm_dmix_i386.h:261 #1 0xb7c44d2b in mix_areas (dmix=0x84a7110, src_areas=0x84a51d0, dst_areas=0x84a6f00, src_ofs=0, dst_ofs=7168, size=1024) at pcm_dmix.c:193 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 #3 0xb7c46680 in snd_pcm_dmix_mmap_commit (pcm=0x84a71e0, offset=0, size=1024) at pcm_dmix.c:813 #4 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a71e0, offset=0, frames=1024) at pcm.c:6501 #5 0xb7c1804e in snd_pcm_plugin_mmap_commit (pcm=0x84a6d18, offset=0, size=1024) at pcm_plugin.c:434 #6 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a6d18, offset=0, frames=1024) at pcm.c:6501 #7 0xb7c17956 in snd_pcm_plugin_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024) at pcm_plugin.c:290 #8 0xb7c074bb in snd1_pcm_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024, func=0xb7c177fc <snd_pcm_plugin_write_areas>) at pcm.c:6667 #9 0xb7c17cd5 in snd_pcm_plugin_writei (pcm=0x84a5390, buffer=0x84ab518, size=1024) at pcm_plugin.c:361 #10 0xb7bff1a7 in _snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm_local.h:516 #11 0xb7bff16c in snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm.c:1258 (gdb) f 2 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 371 pcm_dmix.c: No such file or directory. (gdb) info locals dmix = 0x84a7110 slave_hw_ptr = 234156032 slave_appl_ptr = 7168 slave_size = 1057171456 appl_ptr = 0 size = 1018295296 transfer = 1024 src_areas = 0x84a51d0 dst_areas = 0x84a6f00 (gdb) p *dmix $1 = {type = SND_PCM_TYPE_DMIX, ipc_key = 5678293, ipc_perm = 432, ipc_gid = 17, semid = 32768, shmid = 65536, shmptr = 0xb6859000, spcm = 0x84a7340, appl_ptr = 217121792, last_appl_ptr = 233882624, hw_ptr = 233875428, avail_max = 0, slave_appl_ptr = 234156032, slave_hw_ptr = 234147996, slave_period_size = 1024, slave_buffer_size = 8192, slave_boundary = 1073741824, sync_ptr = 0xb7c4557c <snd_pcm_dmix_sync_ptr>, state = SND_PCM_STATE_RUNNING, trigger_tstamp = {tv_sec = 1392384066, tv_nsec = 8308000}, server = 0, client = 0, comm_fd = 0, hw_fd = 4, timer_fd = { fd = 3, events = 41, revents = 0}, poll_fd = 3, tread = 1, timer_need_poll = 0, timer_event_suspend = 131072, timer_event_resume = 262144, server_fd = 0, server_pid = 0, timer = 0x8496b90, interleaved = 1, slowptr = 1, max_periods = 0, channels = 2, bindings = 0x0, u = {dmix = { shmid_sum = 98305, sum_buffer = 0xb6837000, mix_areas_16 = 0xb7c443a8 <mix_areas_16_smp_mmx>, mix_areas_32 = 0xb7c44415 <mix_areas_32_smp>, mix_areas_24 = 0xb7c4451b <mix_areas_24_smp_cmov>, mix_areas_u8 = 0, remix_areas_16 = 0xb7c44638 <remix_areas_16_smp_mmx>, remix_areas_32 = 0xb7c446a5 <remix_areas_32_smp>, remix_areas_24 = 0xb7c447ab <remix_areas_24_smp_cmov>, remix_areas_u8 = 0}, dsnoop = {<No data fields>}, dshare = {chn_mask = 13151478482154389505}}, server_free = 0} (gdb) p *pcm $2 = {dl_handle = 0x0, name = 0x84a6e48 "dmix:0", type = SND_PCM_TYPE_DMIX, stream = SND_PCM_STREAM_PLAYBACK, mode = 0, poll_fd_count = 1, poll_fd = 3, poll_events = 1, setup = -1, monotonic = 0, access = SND_PCM_ACCESS_MMAP_INTERLEAVED, format = SND_PCM_FORMAT_S32_LE, subformat = SND_PCM_SUBFORMAT_STD, channels = 2, rate = 44100, period_size = 1024, period_time = 23219, periods = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, tstamp_mode = SND_PCM_TSTAMP_NONE, period_step = 1, avail_min = 1024, period_event = 0, start_threshold = 1024, stop_threshold = 2147483647, silence_threshold = 0, silence_size = 0, boundary = 1073741824, info = 2151743747, msbits = 32, rate_num = 44100, rate_den = 1, hw_flags = 0, fifo_size = 0, buffer_size = 2048, buffer_time = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, sample_bits = 32, frame_bits = 64, appl = {master = 0x0, ptr = 0x84a7130, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, hw = { master = 0x0, ptr = 0x84a7138, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, min_align = 1, mmap_rw = 1, mmap_shadow = 0, donot_close = 0, mmap_channels = 0x84a54a0, running_areas = 0x84a51d0, stopped_areas = 0x0, ops = 0xb7c91860, fast_ops = 0xb7c918a0, op_arg = 0x84a71e0, fast_op_arg = 0x84a71e0, private_data = 0x84a7110, async_handlers = {next = 0x84a72e4, prev = 0x84a72e4}} PCM dump: Plug PCM: Linear conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 23219 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 Thanks

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[alsa-devel] [PATCH 00/17 v2] Update locking for snd_soc_dapm_xxxx_pin functions
by Charles Keepax 20 Feb '14

20 Feb '14

This patchset updates the locking around the snd_soc_dapm_xxxx_pin functions. First we add an unlocked version of the functions and add locking to the standard call, this also requires us to remove the few places where these functions are externally locked. Then we patch usages of the functions that look like they are expecting to do an atomic update to do so, using external locking. Thanks, Charles Updates since v1: - Added helper functions to lock/unlock the dapm mutex - Added an unlocked version of snd_soc_dapm_sync, so it can be called under the same mutex lock as the pin functions, this is not required just saves a mutex lock Charles Keepax (17): Input - arizona-haptics: Fix double lock of dapm_mutex ASoC: dapm: Add helpers to lock/unlock DAPM mutex ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions ASoC: dapm: Add unlocked version of snd_soc_dapm_sync ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API drivers/extcon/extcon-arizona.c | 12 --- drivers/input/misc/arizona-haptics.c | 19 ---- include/sound/soc-dapm.h | 9 ++ include/sound/soc.h | 9 ++ sound/soc/codecs/adav80x.c | 17 +++-- sound/soc/codecs/wm5100.c | 12 ++- sound/soc/codecs/wm8962.c | 13 ++- sound/soc/codecs/wm8994.c | 45 ++++++---- sound/soc/codecs/wm8996.c | 9 ++- sound/soc/intel/mfld_machine.c | 65 +++++++++----- sound/soc/omap/ams-delta.c | 45 ++++++---- sound/soc/omap/n810.c | 22 +++-- sound/soc/omap/rx51.c | 22 +++-- sound/soc/pxa/corgi.c | 42 +++++---- sound/soc/pxa/magician.c | 22 +++-- sound/soc/pxa/spitz.c | 51 ++++++----- sound/soc/pxa/tosa.c | 28 ++++--- sound/soc/soc-dapm.c | 160 ++++++++++++++++++++++++++++++--- 18 files changed, 399 insertions(+), 203 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform
by Jarkko Nikula 20 Feb '14

20 Feb '14

Initial implementation of this driver focused only matching SST ACPI ID with single machine driver and same firmware file per platform. It was known restriction to be improved incrementally. This patch is now changing this that SST ACPI ID refers purely to platform specific data which refers to machine drivers on this platform, not vice versa. Matching machine driver is found by looking at ACPI ID which would best match with the driver. Typically this would be the ACPI ID of audio codec but is not tied to it. This patch also changes that DSP firmware name is machine not platform specific. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This goes on top of my "[PATCHv2] ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing". http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/072812.h… --- sound/soc/intel/sst-acpi.c | 84 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 58 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 64154a77d904..c7e36c9ee52f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -26,9 +26,20 @@ #define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 #define SST_LPT_DSP_DMA_SIZE (1024 - 1) +/* Descriptor for SST ASoC machine driver */ +struct sst_acpi_mach { + /* ACPI ID for the matching machine driver. Audio codec for instance */ + const u8 id[ACPI_ID_LEN]; + /* machine driver name */ + const char *drv_name; + /* firmware file name */ + const char *fw_filename; +}; + /* Descriptor for setting up SST platform data */ struct sst_acpi_desc { const char *drv_name; + struct sst_acpi_mach *machines; /* Platform resource indexes. Must set to -1 if not used */ int resindex_lpe_base; int resindex_pcicfg_base; @@ -37,24 +48,17 @@ struct sst_acpi_desc { int resindex_dma_base; /* Unique number identifying the SST core on platform */ int sst_id; - /* firmware file name */ - const char *fw_filename; /* DMA only valid when resindex_dma_base != -1*/ int dma_engine; int dma_size; }; -/* Descriptor for SST ASoC machine driver */ -struct sst_acpi_mach { - const char *drv_name; - struct sst_acpi_desc *res_desc; -}; - struct sst_acpi_priv { struct platform_device *pdev_mach; struct platform_device *pdev_pcm; struct sst_pdata sst_pdata; struct sst_acpi_desc *desc; + struct sst_acpi_mach *mach; }; static void sst_acpi_fw_cb(const struct firmware *fw, void *context) @@ -64,10 +68,11 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; struct sst_acpi_desc *desc = sst_acpi->desc; + struct sst_acpi_mach *mach = sst_acpi->mach; sst_pdata->fw = fw; if (!fw) { - dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename); return; } @@ -83,6 +88,28 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) return; } +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_acpi_mach *sst_acpi_find_machine( + struct sst_acpi_mach *machines) +{ + struct sst_acpi_mach *mach; + bool found = false; + + for (mach = machines; mach->id[0]; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -102,8 +129,13 @@ static int sst_acpi_probe(struct platform_device *pdev) if (!id) return -ENODEV; - mach = (struct sst_acpi_mach *)id->driver_data; - desc = mach->res_desc; + desc = (struct sst_acpi_desc *)id->driver_data; + mach = sst_acpi_find_machine(desc->machines); + if (mach == NULL) { + dev_err(dev, "No matching ASoC machine driver found\n"); + return -ENODEV; + } + sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; @@ -154,7 +186,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return PTR_ERR(sst_acpi->pdev_mach); /* continue SST probing after firmware is loaded */ - ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename, dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); if (ret) platform_device_unregister(sst_acpi->pdev_mach); @@ -175,45 +207,45 @@ static int sst_acpi_remove(struct platform_device *pdev) return 0; } +static struct sst_acpi_mach haswell_machines[] = { + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_haswell_desc = { .drv_name = "haswell-pcm-audio", + .machines = haswell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_LYNX_POINT, - .fw_filename = "intel/IntcSST1.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; +static struct sst_acpi_mach broadwell_machines[] = { + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_broadwell_desc = { .drv_name = "haswell-pcm-audio", + .machines = broadwell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_WILDCAT_POINT, - .fw_filename = "intel/IntcSST2.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; -static struct sst_acpi_mach haswell_mach = { - .drv_name = "haswell-audio", - .res_desc = &sst_acpi_haswell_desc, -}; - -static struct sst_acpi_mach broadwell_mach = { - .drv_name = "broadwell-audio", - .res_desc = &sst_acpi_broadwell_desc, -}; - static struct acpi_device_id sst_acpi_match[] = { - { "INT33C8", (unsigned long)&haswell_mach }, - { "INT3438", (unsigned long)&broadwell_mach }, + { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, + { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, { } }; MODULE_DEVICE_TABLE(acpi, sst_acpi_match); -- 1.8.5.3

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[alsa-devel] [PATCH] ALSA: Export snd_pcm_constraint_mask64()
by Mark Brown 20 Feb '14

20 Feb '14

From: Mark Brown <broonie(a)linaro.org> Allow modules to use it, fixing a build failure when the newly added ADAU1977 driver is built as a module. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- OK to apply this to the ASoC tree? sound/core/pcm_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6630a4e..ce83def 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1240,6 +1240,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); /** * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval -- 1.9.0.rc3

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[alsa-devel] [PATCH] ALSA: hda - Enable front audio jacks on one HP desktop model
by Hui Wang 20 Feb '14

20 Feb '14

The front headphone and mic jackes on a HP desktop model (Vendor Id: 0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this machine has 8 physical ports, 6 of them are routed to rear jackes and all of them work very well, while the remaining 2 ports are routed to front headphone and mic jackes, but the corresponding pin complex node are not defined correctly. After apply this fix, the front audio jackes can work very well. BugLink: https://bugs.launchpad.net/bugs/1282369 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Gerald Yang <gerald.yang(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7311bad..d75f90a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,7 +83,8 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, - STAC_92HD73XX_MODELS + STAC_92HD73XX_MODELS, + STAC_92HD89XX_HP_FRONT_JACK }; enum { @@ -1795,6 +1796,12 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { + { 0x0a, 0x02214030 }, + { 0x0b, 0x02A19010 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1913,6 +1920,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_NO_JD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_no_jd, + }, + [STAC_92HD89XX_HP_FRONT_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_front_jack_pin_configs, } }; @@ -1973,6 +1984,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, + "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ }; -- 1.8.1.2

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