From dd37f8e8659bc617c3f2a84e007a4824ccdac458 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 30 May 2010 01:17:03 -0400 Subject: ALSA: hda: Use LPIB for an ASUS device BugLink: https://launchpad.net/bugs/465942 Symptom: On the reporter's ASUS device, using PulseAudio in Ubuntu 10.04 LTS results in the PA daemon crashing shortly after attempting to select capture or to configure the audio hardware profile. Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or Linux 2.6.34, adjust the HDA device's capture volume with PulseAudio. Resolution: add SSID for this machine to the position_fix quirk table, explicitly specifying the LPIB method. Reported-and-Tested-By: Irihapeti Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dc79564..7c54a40 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2288,6 +2288,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), -- cgit v1.1 From 26fd74fc01991a18f0e3bd54f8b1b75945ee3dbb Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 30 May 2010 09:55:23 -0400 Subject: ALSA: hda: Use mb31 quirk for an iMac model BugLink: https://launchpad.net/bugs/542550 Symptom: On the reporter's iMac, in Ubuntu 10.04 LTS neither playback nor capture appear audible out-of-the-box. Test case: Boot from an Ubuntu 10.04 LTS live cd or from an installed configuration and attempt to play or capture audio. Resolution: Specify the mb31 quirk for this machine in the codec SSID table. Reported-and-Tested-By: f3a97 Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17d4548..d792cdd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9476,6 +9476,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), -- cgit v1.1 From b90c076424da8166797bdc34187660fd0124f530 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 30 May 2010 19:31:41 -0400 Subject: ALSA: hda: Use LPIB for another mainboard BugLink: https://launchpad.net/bugs/580749 Symptom: on the original reporter's VIA VT1708-based board, the PulseAudio daemon dies shortly after the user attempts to play an audio file. Test case: boot from Ubuntu 10.04 LTS live cd; attempt to play an audio file. Resolution: add SSID for the original reporter's hardware to the position_fix quirk table, explicitly specifying the LPIB method. Reported-and-Tested-By: Harald Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7c54a40..1640005 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2297,6 +2297,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} -- cgit v1.1 From dcbe7bcfa32c5bc4f9bb6c75d4d41bb4db8c36fc Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 13:35:36 +0200 Subject: ALSA: usb-audio: UAC2: clean up parsing of bmaControls Introduce two new static inline functions for a more readable parsing of UAC2 bmaControls. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 03ce971..43d6417 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1188,9 +1188,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void for (j = 0; j < channels; j++) { unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); - if (mask & (1 << (i * 2))) { + if (uac2_control_is_readable(mask, i)) { ch_bits |= (1 << j); - if (~mask & (1 << ((i * 2) + 1))) + if (!uac2_control_is_writeable(mask, i)) ch_read_only |= (1 << j); } } @@ -1198,9 +1198,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* FIXME: the whole unit is read-only if any of the channels is marked read-only */ if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only); - if (master_bits & (1 << i * 2)) + if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, - ~master_bits & (1 << ((i * 2) + 1))); + !uac2_control_is_writeable(master_bits, i)); } } -- cgit v1.1 From a6a3325913efbe35a10e87fd3e9c3ce621fd32c7 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 13:35:37 +0200 Subject: ALSA: usb-audio: support partially write-protected UAC2 controls So far, UAC2 controls are marked read-only if any of the channels are marked read-only in the descriptors. Change this behaviour and - mark them writeable unless all channels are read-only - store the read-only mask in usb_mixer_elem_info and - check the mask again in set_cur_mix_value(), and bail out for write-protected channels. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 30 ++++++++++++++++++++++++------ sound/usb/mixer.h | 2 ++ 2 files changed, 26 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 43d6417..9149a84 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -462,6 +462,16 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; + unsigned int read_only = (channel == 0) ? + cval->master_readonly : + cval->ch_readonly & (1 << (channel - 1)); + + if (read_only) { + snd_printdd(KERN_INFO "%s(): channel %d of control %d is read_only\n", + __func__, channel, cval->control); + return 0; + } + err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) @@ -958,7 +968,7 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) static void build_feature_ctl(struct mixer_build *state, void *raw_desc, unsigned int ctl_mask, int control, struct usb_audio_term *iterm, int unitid, - int read_only) + int readonly_mask) { struct uac_feature_unit_descriptor *desc = raw_desc; unsigned int len = 0; @@ -989,20 +999,25 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->control = control; cval->cmask = ctl_mask; cval->val_type = audio_feature_info[control-1].type; - if (ctl_mask == 0) + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ - else { + cval->master_readonly = readonly_mask; + } else { int i, c = 0; for (i = 0; i < 16; i++) if (ctl_mask & (1 << i)) c++; cval->channels = c; + cval->ch_readonly = readonly_mask; } /* get min/max values */ get_min_max(cval, 0); - if (read_only) + /* if all channels in the mask are marked read-only, make the control + * read-only. set_cur_mix_value() will check the mask again and won't + * issue write commands to read-only channels. */ + if (cval->channels == readonly_mask) kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); else kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); @@ -1195,9 +1210,12 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } } - /* FIXME: the whole unit is read-only if any of the channels is marked read-only */ + /* NOTE: build_feature_ctl() will mark the control read-only if all channels + * are marked read-only in the descriptors. Otherwise, the control will be + * reported as writeable, but the driver will not actually issue a write + * command for read-only channels */ if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only); + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, !uac2_control_is_writeable(master_bits, i)); diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 1301238..a7cf100 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -34,6 +34,8 @@ struct usb_mixer_elem_info { unsigned int id; unsigned int control; /* CS or ICN (high byte) */ unsigned int cmask; /* channel mask bitmap: 0 = master */ + unsigned int ch_readonly; + unsigned int master_readonly; int channels; int val_type; int min, max, res; -- cgit v1.1 From 79f920fbff566ffc9de44111eb1456a3cef310f0 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 14:51:31 +0200 Subject: ALSA: usb-audio: parse clock topology of UAC2 devices Audio devices which comply to the UAC2 standard can export complex clock topologies in its descriptors and set up links between them. The entities that are defined are - clock sources, which define the end-leafs. - clock selectors, which act as switch to select one out of many possible clocks sources. - clock multipliers, which have an input clock source, and act as clock source again. They can be used to derive one clock from another. All sample rate changes, clock validity queries and the like must go to clock source elements, while clock selectors and multipliers can be used as terminal clock source. The following patch adds a parser for these elements and functions to iterate over the tree and find the leaf nodes (clock sources). The samplerate set functions were moved to the new clock.c file. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/Makefile | 3 +- sound/usb/card.c | 18 +-- sound/usb/card.h | 1 + sound/usb/clock.c | 311 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/clock.h | 12 ++ sound/usb/endpoint.c | 57 +++++++++- sound/usb/format.c | 16 ++- sound/usb/pcm.c | 98 +--------------- sound/usb/usbaudio.h | 5 +- 9 files changed, 396 insertions(+), 125 deletions(-) create mode 100644 sound/usb/clock.c create mode 100644 sound/usb/clock.h (limited to 'sound') diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e7ac7f4..1e362bf 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,7 +11,8 @@ snd-usb-audio-objs := card.o \ endpoint.o \ urb.o \ pcm.o \ - helper.o + helper.o \ + clock.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/card.c b/sound/usb/card.c index da1346b..7a8ac1d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -236,7 +236,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } case UAC_VERSION_2: { - struct uac_clock_source_descriptor *cs; struct usb_interface_assoc_descriptor *assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc; @@ -245,21 +244,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) return -EINVAL; } - /* FIXME: for now, we expect there is at least one clock source - * descriptor and we always take the first one. - * We should properly support devices with multiple clock sources, - * clock selectors and sample rate conversion units. */ - - cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, UAC2_CLOCK_SOURCE); - - if (!cs) { - snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); - return -EINVAL; - } - - chip->clock_id = cs->bClockID; - for (i = 0; i < assoc->bInterfaceCount; i++) { int intf = assoc->bFirstInterface + i; @@ -481,6 +465,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } + chip->ctrl_intf = alts; + if (err > 0) { /* create normal USB audio interfaces */ if (snd_usb_create_streams(chip, ifnum) < 0 || diff --git a/sound/usb/card.h b/sound/usb/card.h index ed92420..1febf2f 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -25,6 +25,7 @@ struct audioformat { unsigned int rate_min, rate_max; /* min/max rates */ unsigned int nr_rates; /* number of rate table entries */ unsigned int *rate_table; /* rate table */ + unsigned char clock; /* associated clock */ }; struct snd_usb_substream; diff --git a/sound/usb/clock.c b/sound/usb/clock.c new file mode 100644 index 0000000..b7aadd6 --- /dev/null +++ b/sound/usb/clock.c @@ -0,0 +1,311 @@ +/* + * Clock domain and sample rate management functions + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "usbaudio.h" +#include "card.h" +#include "midi.h" +#include "mixer.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "helper.h" +#include "debug.h" +#include "pcm.h" +#include "urb.h" +#include "format.h" + +static struct uac_clock_source_descriptor * + snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_source_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_SOURCE))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static struct uac_clock_selector_descriptor * + snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_selector_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_SELECTOR))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static struct uac_clock_multiplier_descriptor * + snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_multiplier_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_MULTIPLIER))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) +{ + unsigned char buf; + int ret; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), + UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + UAC2_CX_CLOCK_SELECTOR << 8, selector_id << 8, + &buf, sizeof(buf), 1000); + + if (ret < 0) + return ret; + + return buf; +} + +static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) +{ + int err; + unsigned char data; + struct usb_device *dev = chip->dev; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, source_id << 8, + &data, sizeof(data), 1000); + + if (err < 0) { + snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return err; + } + + return !!data; +} + +/* Try to find the clock source ID of a given clock entity */ + +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id, unsigned long *visited) +{ + struct uac_clock_source_descriptor *source; + struct uac_clock_selector_descriptor *selector; + struct uac_clock_multiplier_descriptor *multiplier; + + entity_id &= 0xff; + + if (test_and_set_bit(entity_id, visited)) { + snd_printk(KERN_WARNING + "%s(): recursive clock topology detected, id %d.\n", + __func__, entity_id); + return -EINVAL; + } + + /* first, see if the ID we're looking for is a clock source already */ + source = snd_usb_find_clock_source(host_iface, entity_id); + if (source) + return source->bClockID; + + selector = snd_usb_find_clock_selector(host_iface, entity_id); + if (selector) { + int ret; + + /* the entity ID we are looking for is a selector. + * find out what it currently selects */ + ret = uac_clock_selector_get_val(chip, selector->bClockID); + if (ret < 0) + return ret; + + if (ret > selector->bNrInPins || ret < 1) { + printk(KERN_ERR + "%s(): selector reported illegal value, id %d, ret %d\n", + __func__, selector->bClockID, ret); + + return -EINVAL; + } + + return __uac_clock_find_source(chip, host_iface, + selector->baCSourceID[ret-1], + visited); + } + + /* FIXME: multipliers only act as pass-thru element for now */ + multiplier = snd_usb_find_clock_multiplier(host_iface, entity_id); + if (multiplier) + return __uac_clock_find_source(chip, host_iface, + multiplier->bCSourceID, visited); + + return -EINVAL; +} + +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id) +{ + DECLARE_BITMAP(visited, 256); + memset(visited, 0, sizeof(visited)); + return __uac_clock_find_source(chip, host_iface, entity_id, visited); +} + +static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned int ep; + unsigned char data[3]; + int err, crate; + + ep = get_endpoint(alts, 0)->bEndpointAddress; + + /* if endpoint doesn't have sampling rate control, bail out */ + if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { + snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", + dev->devnum, iface, fmt->altsetting); + return 0; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", + dev->devnum, iface, fmt->altsetting, rate, ep); + return err; + } + + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", + dev->devnum, iface, fmt->altsetting, ep); + return 0; /* some devices don't support reading */ + } + + crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (crate != rate) { + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + // runtime->rate = crate; + } + + return 0; +} + +static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned char data[4]; + int err, crate; + int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fmt->clock); + + if (clock < 0) + return clock; + + if (!uac_clock_source_is_valid(chip, clock)) { + snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n", + dev->devnum, iface, fmt->altsetting, clock); + return -ENXIO; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + data[3] = rate >> 24; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", + dev->devnum, iface, fmt->altsetting, rate); + return err; + } + + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (crate != rate) + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + + return 0; +} + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + + switch (altsd->bInterfaceProtocol) { + case UAC_VERSION_1: + return set_sample_rate_v1(chip, iface, alts, fmt, rate); + + case UAC_VERSION_2: + return set_sample_rate_v2(chip, iface, alts, fmt, rate); + } + + return -EINVAL; +} + diff --git a/sound/usb/clock.h b/sound/usb/clock.h new file mode 100644 index 0000000..beb2536 --- /dev/null +++ b/sound/usb/clock.h @@ -0,0 +1,12 @@ +#ifndef __USBAUDIO_CLOCK_H +#define __USBAUDIO_CLOCK_H + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate); + +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id); + +#endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 28ee1ce..9593b91 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -190,6 +190,38 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, return attributes; } +static struct uac2_input_terminal_descriptor * + snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_input_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_INPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +static struct uac2_output_terminal_descriptor * + snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_output_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_OUTPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -199,7 +231,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) int i, altno, err, stream; int format = 0, num_channels = 0; struct audioformat *fp = NULL; - int num, protocol; + int num, protocol, clock = 0; struct uac_format_type_i_continuous_descriptor *fmt; dev = chip->dev; @@ -263,6 +295,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } case UAC_VERSION_2: { + struct uac2_input_terminal_descriptor *input_term; + struct uac2_output_terminal_descriptor *output_term; struct uac_as_header_descriptor_v2 *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -281,7 +315,25 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num_channels = as->bNrChannels; format = le32_to_cpu(as->bmFormats); - break; + /* lookup the terminal associated to this interface + * to extract the clock */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + break; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + break; + } + + snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", + dev->devnum, iface_no, altno, as->bTerminalLink); + continue; } default: @@ -338,6 +390,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); + fp->clock = clock; /* some quirks for attributes here */ diff --git a/sound/usb/format.c b/sound/usb/format.c index fe29d61..5367cd1 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -29,6 +29,7 @@ #include "quirks.h" #include "helper.h" #include "debug.h" +#include "clock.h" /* * parse the audio format type I descriptor @@ -215,15 +216,17 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int i, nr_rates, data_size, ret = 0; + int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock); /* get the number of sample rates first by only fetching 2 bytes */ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, tmp, sizeof(tmp), 1000); if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); goto err; } @@ -237,12 +240,13 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, /* now get the full information */ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, data_size, 1000); + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, + data, data_size, 1000); if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", + __func__, clock); ret = -EINVAL; goto err_free; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 056587d..4568298 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -31,6 +31,7 @@ #include "urb.h" #include "helper.h" #include "pcm.h" +#include "clock.h" /* * return the current pcm pointer. just based on the hwptr_done value. @@ -181,103 +182,6 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, return -EINVAL; } -static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_device *dev = chip->dev; - unsigned int ep; - unsigned char data[3]; - int err, crate; - - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint doesn't have sampling rate control, bail out */ - if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { - snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", - dev->devnum, iface, fmt->altsetting); - return 0; - } - - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", - dev->devnum, iface, fmt->altsetting, rate, ep); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", - dev->devnum, iface, fmt->altsetting, ep); - return 0; /* some devices don't support reading */ - } - crate = data[0] | (data[1] << 8) | (data[2] << 16); - if (crate != rate) { - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - // runtime->rate = crate; - } - - return 0; -} - -static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_device *dev = chip->dev; - unsigned char data[4]; - int err, crate; - - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - data[3] = rate >> 24; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", - dev->devnum, iface, fmt->altsetting, rate); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", - dev->devnum, iface, fmt->altsetting); - return err; - } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - - return 0; -} - -int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - - switch (altsd->bInterfaceProtocol) { - case UAC_VERSION_1: - return set_sample_rate_v1(chip, iface, alts, fmt, rate); - - case UAC_VERSION_2: - return set_sample_rate_v2(chip, iface, alts, fmt, rate); - } - - return -EINVAL; -} - /* * find a matching format and set up the interface */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 06ebf24..24d3319 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -40,9 +40,6 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; - /* for audio class v2 */ - int clock_id; - struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; @@ -53,6 +50,8 @@ struct snd_usb_audio { int setup; /* from the 'device_setup' module param */ int nrpacks; /* from the 'nrpacks' module param */ int async_unlink; /* from the 'async_unlink' module param */ + + struct usb_host_interface *ctrl_intf; /* the audio control interface */ }; /* -- cgit v1.1 From 65f25da44b51f55e3a74301c25f29263be2bf1ba Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 13:35:41 +0200 Subject: ALSA: usb-audio: unify constants from specification Move more definitions from private enums to appropriate header files. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 99 +++++++++++++++++--------------------------------- sound/usb/mixer_maps.c | 4 +- 2 files changed, 35 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9149a84..2442819 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -78,39 +78,6 @@ enum { USB_MIXER_U16, }; -enum { - USB_PROC_UPDOWN = 1, - USB_PROC_UPDOWN_SWITCH = 1, - USB_PROC_UPDOWN_MODE_SEL = 2, - - USB_PROC_PROLOGIC = 2, - USB_PROC_PROLOGIC_SWITCH = 1, - USB_PROC_PROLOGIC_MODE_SEL = 2, - - USB_PROC_3DENH = 3, - USB_PROC_3DENH_SWITCH = 1, - USB_PROC_3DENH_SPACE = 2, - - USB_PROC_REVERB = 4, - USB_PROC_REVERB_SWITCH = 1, - USB_PROC_REVERB_LEVEL = 2, - USB_PROC_REVERB_TIME = 3, - USB_PROC_REVERB_DELAY = 4, - - USB_PROC_CHORUS = 5, - USB_PROC_CHORUS_SWITCH = 1, - USB_PROC_CHORUS_LEVEL = 2, - USB_PROC_CHORUS_RATE = 3, - USB_PROC_CHORUS_DEPTH = 4, - - USB_PROC_DCR = 6, - USB_PROC_DCR_SWITCH = 1, - USB_PROC_DCR_RATIO = 2, - USB_PROC_DCR_MAX_AMP = 3, - USB_PROC_DCR_THRESHOLD = 4, - USB_PROC_DCR_ATTACK = 5, - USB_PROC_DCR_RELEASE = 6, -}; /*E-mu 0202(0404) eXtension Unit(XU) control*/ enum { @@ -980,7 +947,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, control++; /* change from zero-based to 1-based value */ - if (control == UAC_GRAPHIC_EQUALIZER_CONTROL) { + if (control == UAC_FU_GRAPHIC_EQUALIZER) { /* FIXME: not supported yet */ return; } @@ -1036,8 +1003,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kctl->id.name, sizeof(kctl->id.name)); switch (control) { - case UAC_MUTE_CONTROL: - case UAC_VOLUME_CONTROL: + case UAC_FU_MUTE: + case UAC_FU_VOLUME: /* determine the control name. the rule is: * - if a name id is given in descriptor, use it. * - if the connected input can be determined, then use the name @@ -1064,9 +1031,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = append_ctl_name(kctl, " Playback"); } } - append_ctl_name(kctl, control == UAC_MUTE_CONTROL ? + append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); - if (control == UAC_VOLUME_CONTROL) { + if (control == UAC_FU_VOLUME) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | @@ -1165,7 +1132,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void snd_printk(KERN_INFO "usbmixer: master volume quirk for PCM2702 chip\n"); /* disable non-functional volume control */ - master_bits &= ~UAC_FU_VOLUME; + master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); break; } if (channels > 0) @@ -1410,51 +1377,51 @@ struct procunit_info { }; static struct procunit_value_info updown_proc_info[] = { - { USB_PROC_UPDOWN_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 }, + { UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; static struct procunit_value_info prologic_proc_info[] = { - { USB_PROC_PROLOGIC_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 }, + { UAC_DP_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_DP_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; static struct procunit_value_info threed_enh_proc_info[] = { - { USB_PROC_3DENH_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_3DENH_SPACE, "Spaciousness", USB_MIXER_U8 }, + { UAC_3D_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_3D_SPACE, "Spaciousness", USB_MIXER_U8 }, { 0 } }; static struct procunit_value_info reverb_proc_info[] = { - { USB_PROC_REVERB_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_REVERB_LEVEL, "Level", USB_MIXER_U8 }, - { USB_PROC_REVERB_TIME, "Time", USB_MIXER_U16 }, - { USB_PROC_REVERB_DELAY, "Delay", USB_MIXER_U8 }, + { UAC_REVERB_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_REVERB_LEVEL, "Level", USB_MIXER_U8 }, + { UAC_REVERB_TIME, "Time", USB_MIXER_U16 }, + { UAC_REVERB_FEEDBACK, "Feedback", USB_MIXER_U8 }, { 0 } }; static struct procunit_value_info chorus_proc_info[] = { - { USB_PROC_CHORUS_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_CHORUS_LEVEL, "Level", USB_MIXER_U8 }, - { USB_PROC_CHORUS_RATE, "Rate", USB_MIXER_U16 }, - { USB_PROC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 }, + { UAC_CHORUS_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_CHORUS_LEVEL, "Level", USB_MIXER_U8 }, + { UAC_CHORUS_RATE, "Rate", USB_MIXER_U16 }, + { UAC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 }, { 0 } }; static struct procunit_value_info dcr_proc_info[] = { - { USB_PROC_DCR_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_DCR_RATIO, "Ratio", USB_MIXER_U16 }, - { USB_PROC_DCR_MAX_AMP, "Max Amp", USB_MIXER_S16 }, - { USB_PROC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 }, - { USB_PROC_DCR_ATTACK, "Attack Time", USB_MIXER_U16 }, - { USB_PROC_DCR_RELEASE, "Release Time", USB_MIXER_U16 }, + { UAC_DCR_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_DCR_RATE, "Ratio", USB_MIXER_U16 }, + { UAC_DCR_MAXAMPL, "Max Amp", USB_MIXER_S16 }, + { UAC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 }, + { UAC_DCR_ATTACK_TIME, "Attack Time", USB_MIXER_U16 }, + { UAC_DCR_RELEASE_TIME, "Release Time", USB_MIXER_U16 }, { 0 } }; static struct procunit_info procunits[] = { - { USB_PROC_UPDOWN, "Up Down", updown_proc_info }, - { USB_PROC_PROLOGIC, "Dolby Prologic", prologic_proc_info }, - { USB_PROC_3DENH, "3D Stereo Extender", threed_enh_proc_info }, - { USB_PROC_REVERB, "Reverb", reverb_proc_info }, - { USB_PROC_CHORUS, "Chorus", chorus_proc_info }, - { USB_PROC_DCR, "DCR", dcr_proc_info }, + { UAC_PROCESS_UP_DOWNMIX, "Up Down", updown_proc_info }, + { UAC_PROCESS_DOLBY_PROLOGIC, "Dolby Prologic", prologic_proc_info }, + { UAC_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", threed_enh_proc_info }, + { UAC_PROCESS_REVERB, "Reverb", reverb_proc_info }, + { UAC_PROCESS_CHORUS, "Chorus", chorus_proc_info }, + { UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info }, { 0 }, }; /* @@ -1542,7 +1509,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->channels = 1; /* get min/max values */ - if (type == USB_PROC_UPDOWN && cval->control == USB_PROC_UPDOWN_MODE_SEL) { + if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) { __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol); /* FIXME: hard-coded */ cval->min = 1; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index d93fc89..f1324c4 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -85,8 +85,8 @@ static struct usbmix_name_map extigy_map[] = { /* 16: MU (w/o controls) */ { 17, NULL, 1 }, /* DISABLED: PU-switch (any effect?) */ { 17, "Channel Routing", 2 }, /* PU: mode select */ - { 18, "Tone Control - Bass", UAC_BASS_CONTROL }, /* FU */ - { 18, "Tone Control - Treble", UAC_TREBLE_CONTROL }, /* FU */ + { 18, "Tone Control - Bass", UAC_FU_BASS }, /* FU */ + { 18, "Tone Control - Treble", UAC_FU_TREBLE }, /* FU */ { 18, "Master Playback" }, /* FU; others */ /* 19: OT speaker */ /* 20: OT headphone */ -- cgit v1.1 From 2e0281d15c220d0a81c45c73872aa08d2f3ae3ef Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 13:35:42 +0200 Subject: ALSA: usb-audio: add UAC2 sepecific Feature Unit controls The bits to enable them are always 0 for UAC1 devices, so no additional checks are required. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2442819..8be6bf2 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -686,16 +686,20 @@ struct usb_feature_control_info { }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, + { "Mute", USB_MIXER_INV_BOOLEAN }, + { "Volume", USB_MIXER_S16 }, { "Tone Control - Bass", USB_MIXER_S8 }, { "Tone Control - Mid", USB_MIXER_S8 }, { "Tone Control - Treble", USB_MIXER_S8 }, { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Auto Gain Control", USB_MIXER_BOOLEAN }, + { "Delay Control", USB_MIXER_U16 }, + { "Bass Boost", USB_MIXER_BOOLEAN }, + { "Loudness", USB_MIXER_BOOLEAN }, + /* UAC2 specific */ + { "Input Gain Control", USB_MIXER_U16 }, + { "Input Gain Pad Control", USB_MIXER_BOOLEAN }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; -- cgit v1.1 From 67e1daa0bb30eda6ec5add27c3abf4536030f5a6 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 13:35:43 +0200 Subject: ALSA: usb-audio: clean up find_audio_control_unit() Use a struct to parse the audio units, and return usable descriptors for all types. There's no need to limit the result set, except for some kind of sanity check. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 8be6bf2..cb345360 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -165,22 +165,24 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, /* * find an audio control unit with the given unit id - * this doesn't return any clock related units, so they need to be handled elsewhere */ static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit) { - unsigned char *p; + /* we just parse the header */ + struct uac_feature_unit_descriptor *hdr = NULL; - p = NULL; - while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, - USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC2_EXTENSION_UNIT_V2 && p[3] == unit) - return p; + while ((hdr = snd_usb_find_desc(state->buffer, state->buflen, hdr, + USB_DT_CS_INTERFACE)) != NULL) { + if (hdr->bLength >= 4 && + hdr->bDescriptorSubtype >= UAC_INPUT_TERMINAL && + hdr->bDescriptorSubtype <= UAC2_SAMPLE_RATE_CONVERTER && + hdr->bUnitID == unit) + return hdr; } + return NULL; } - /* * copy a string with the given id */ -- cgit v1.1 From 09414207d4daab8c4990bface3a79fdba3474bec Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 31 May 2010 13:35:44 +0200 Subject: ALSA: usb-audio: export UAC2 clock selectors as mixer controls The UAC2 clock selectors are fortunately compatible with UAC1 audio selector units, so we can simply reuse the same approach to get all the linked units. Requests to this control need a different CS value though. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 42 +++++++++++++++++++++++++++++++++++------- 1 file changed, 35 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index cb345360..a060d00 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -313,8 +313,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v buf, sizeof(buf), 1000); if (ret < 0) { - snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); + snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", + request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); return ret; } @@ -610,6 +610,7 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm */ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { + int err; void *p1; memset(term, 0, sizeof(*term)); @@ -630,6 +631,11 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->channels = d->bNrChannels; term->chconfig = le32_to_cpu(d->bmChannelConfig); term->name = d->iTerminal; + + /* call recursively to get the clock selectors */ + err = check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; } return 0; case UAC_FEATURE_UNIT: { @@ -646,7 +652,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->name = uac_mixer_unit_iMixer(d); return 0; } - case UAC_SELECTOR_UNIT: { + case UAC_SELECTOR_UNIT: + case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ if (check_input_term(state, d->baSourceID[0], term) < 0) @@ -669,6 +676,13 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->name = uac_processing_unit_iProcessing(d, state->mixer->protocol); return 0; } + case UAC2_CLOCK_SOURCE: { + struct uac_clock_source_descriptor *d = p1; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = d->iClockSource; + return 0; + } default: return -ENODEV; } @@ -1610,7 +1624,7 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, err; - err = get_cur_ctl_value(cval, 0, &val); + err = get_cur_ctl_value(cval, cval->control << 8, &val); if (err < 0) { if (cval->mixer->ignore_ctl_error) { ucontrol->value.enumerated.item[0] = 0; @@ -1629,7 +1643,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, oval, err; - err = get_cur_ctl_value(cval, 0, &oval); + err = get_cur_ctl_value(cval, cval->control << 8, &oval); if (err < 0) { if (cval->mixer->ignore_ctl_error) return 0; @@ -1638,7 +1652,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ val = ucontrol->value.enumerated.item[0]; val = get_abs_value(cval, val); if (val != oval) { - set_cur_ctl_value(cval, 0, val); + set_cur_ctl_value(cval, cval->control << 8, val); return 1; } return 0; @@ -1720,6 +1734,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void cval->res = 1; cval->initialized = 1; + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) + cval->control = UAC2_CX_CLOCK_SELECTOR; + else + cval->control = 0; + namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); if (! namelist) { snd_printk(KERN_ERR "cannot malloc\n"); @@ -1769,7 +1788,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void if (! len) strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); - if ((state->oterm.type & 0xff00) == 0x0100) + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) + append_ctl_name(kctl, " Clock Source"); + else if ((state->oterm.type & 0xff00) == 0x0100) append_ctl_name(kctl, " Capture Source"); else append_ctl_name(kctl, " Playback Source"); @@ -1803,10 +1824,12 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: + case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: + case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); @@ -1903,6 +1926,11 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) return err; + + /* for UAC2, use the same approach to also add the clock selectors */ + err = parse_audio_unit(&state, desc->bCSourceID); + if (err < 0) + return err; } } -- cgit v1.1 From fc9cbe3998ea23a0658c97159c35765c98eafa37 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Mon, 31 May 2010 10:49:54 +0200 Subject: ASoC: Add missing Kconfig entry for Phytec boards Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index eba9b9d..de508a5 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -19,3 +19,12 @@ config SND_MXC_SOC_WM1133_EV1 help Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. + +config SND_SOC_PHYCORE_AC97 + tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" + depends on MACH_PCM043 || MACH_PCA100 + select SND_MXC_SOC_SSI + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on Phytec phyCORE + and phyCARD boards in AC97 mode -- cgit v1.1 From 29512c95b5e2f0f245bfa4975ccae6c3449d4dd2 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Mon, 31 May 2010 14:19:50 +0200 Subject: ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1 Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index de508a5..252defe 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -13,7 +13,7 @@ config SND_MXC_SOC_SSI config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" - depends on SND_IMX_SOC && EXPERIMENTAL + depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 select SND_MXC_SOC_SSI help -- cgit v1.1 From 9f75c1b12c5ef392ddcea575b13560842c28b1b3 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 30 May 2010 13:08:41 -0400 Subject: ALSA: hda: Use LPIB for ASUS M2V BugLink: https://launchpad.net/bugs/587546 Symptom: On the reporter's ASUS M2V, using PulseAudio in Ubuntu 10.04 LTS results in the PA daemon crashing shortly after attempting playback of an audio file. Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or Linux 2.6.34, attempt playback of an audio file while PulseAudio is active. Resolution: add SSID for this machine to the position_fix quirk table, explicitly specifying the LPIB method. Reported-and-Tested-By: D Tangman Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1640005..e42ab99 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2289,6 +2289,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), -- cgit v1.1 From 21896bc010c17e5ac58951e771496ec2fb1051ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Jun 2010 12:08:37 +0200 Subject: ALSA: asihpi - Fix uninitialized variable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Initialize prev_ctl properly before reference: sound/pci/asihpi/asihpi.c: In function ‘snd_card_asihpi_mixer_new’: sound/pci/asihpi/asihpi.c:2568:30: warning: ‘prev_ctl.dst_node_index’ may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index f74c737..1db586a 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2578,6 +2578,9 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) if (err) return -err; + memset(&prev_ctl, 0, sizeof(prev_ctl)); + prev_ctl.control_type = -1; + for (idx = 0; idx < 2000; idx++) { err = hpi_mixer_get_control_by_index( ss, asihpi->h_mixer, -- cgit v1.1 From edb39935c8b19fcd9a8f619d0bc1e9d04594cd2b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 2 Jun 2010 13:29:17 +0200 Subject: ALSA: hda-intel - fix wallclk variable update and condition This patch fixes thinko introduced in "last minutes" before commiting of the last wallclk patch. It also fixes the condition checking if the first period after last wallclk update is processed. There is a little rounding error in period_wallclk. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dc79564..af701a8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1913,11 +1913,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return -1; /* this shouldn't happen! */ - if (wallclk <= azx_dev->period_wallclk && + if (wallclk < (azx_dev->period_wallclk * 5) / 4 && pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) /* NG - it's below the first next period boundary */ return bdl_pos_adj[chip->dev_index] ? 0 : -1; - azx_dev->start_wallclk = wallclk; + azx_dev->start_wallclk += wallclk; return 1; /* OK, it's fine */ } -- cgit v1.1