May 2010 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ALSA, hda: Add support for Lenovo ThinkPad Edge 13 in conexant codec
by Takashi Iwai 28 May '10

28 May '10

At Thu, 27 May 2010 18:06:56 +0200, Andreas Herrmann wrote: > > From: Andreas Herrmann <andreas.herrmann3(a)amd.com> > > On a Thinkpad Edge 13 I had the problem that speakers played sound > although headphones were plugged. > > Using "model=olpc-xo-1_5" solved this problem. So it seems sensible to > add this quirk. (Using other models, like ideabad did not fix it.) > > Signed-off-by: Andreas Herrmann <andreas.herrmann3(a)amd.com> > --- > sound/pci/hda/patch_conexant.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > I don't know whether adding this quirk is the right/complete solution > to support sound on Thinkpad Edge. But at least it solves this > annoying problem. Well, Jerone Young already sent a quirk patch but he took model=ideapd. Could you guys check which one is really better? thanks, Takashi > > Please apply. > > > Thanks, > > Andreas > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index feabb44..f85f072 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -2848,6 +2848,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { > SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), > + SND_PCI_QUIRK(0x17aa, 0x21b3, "Thnkpad Edge 13", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), > {} > }; > -- > 1.6.4.4 >

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Re: [alsa-devel] [PATCH 3/4] ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
by Takashi Iwai 28 May '10

28 May '10

At Wed, 26 May 2010 13:46:57 -0700, Greg KH wrote: > > On Wed, May 26, 2010 at 06:17:46PM +0200, Daniel Mack wrote: > > On Wed, May 26, 2010 at 06:11:38PM +0200, Daniel Mack wrote: > > > UAC2 devices have their information about pitch control stored in a > > > different field. Parse it, and emulate the bits for a v1 device. > > > > > > A new struct uac2_iso_endpoint_descriptor is added. > > > > > > > > Signed-off-by: Daniel Mack <daniel(a)caiaq.de> > > > --- > > > include/linux/usb/audio-v2.h | 16 ++++++++++++ > > > > Greg, I copied you because of this patch, just to let you know we're > > having fun in your area of responsibility again :) > > Heh, no problem at all. > > Takashi, feel free to add an: > Acked-by: Greg Kroah-Hartman <gregkh(a)suse.de> > to this patch when you take it. OK, I put this to my tree. Will be included in the next pull request in this week. thanks, Takashi

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Sony VPCS11V9E
by Daniel Chen 28 May '10

28 May '10

Hi, Attached is a patch against sound-2.6 master HEAD that resolves stuttering audio on the original reporter's hardware. Please apply. This patch is applicable to all stable trees of 2.6.3x. Best, -Dan

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[alsa-devel] usbaudio: Support for USB audio v2.0 devices
by Daniel Mack 27 May '10

27 May '10

Hi, I've been working on support for USB devices compliant to audio class v2.0 and would like to share what I got. First problem is that they literally changed every single descriptor type by adding or removing fields or changing their width, respectively. Even worse, some enum values were dropped and reassigned so their numerical value has a different meaning now. My first step was to clean up most of the existing descriptor parsers by moving away from anonymous byte array access to real structs so we can see which fields we're actually dereferencing. In a second step, I added definitions for the replacement descriptors and use them at appropriate places. The bInterfaceProtocol field must always be set correctly of course, otherwise we'll parse garbage. Other things that have changed from v1.0 to v2.0 are: * The number of streaming interfaces is now reported by a standard interface association descriptor. The old approach using a proprietary descriptor extension is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the FORMAT_TYPE descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than as a fixed value. For now, this is worked around by choosing just one of them. Eventually, the code should propagate the full ability of the device. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed as argument to the class request command. If can have more than one though and the host driver is in charge to choose the right one. There are also new types for clock selectors and sample rate converters that a device might report. This is all unsupported at the moment. We only parse the first CLOCK_SOURCE and take this one. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. With the following patches applied, a v2.0 device is recognized and reported as ALSA device. I can even hear at least some kind of music, but the sample rate setting does not yet seem to be correct. Also, the existing mixer code does not work at all for v2.0 devices. This will be quite some work to support that and I fear the whole thing must be reimplemented for v2.0 specifically. For now, I worked around that by not parsing them at all but bailing out very early. However, I would like to make sure the changes I made so far won't cause any regression for v1.0 devices, and so it would be good to see them applied as a preliminary support. I'll continue working on this and send more updates along. Let me know what you think. Thanks, Daniel

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[PATCH] ALSA: usb-audio: fix feature unit parser for UAC2
by Daniel Mack 27 May '10

27 May '10

Fix a small off-by-one bug which causes the feature unit to announce a wrong number of channels. This leads to illegal requests sent to the firmware eventually. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/mixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 97dd176..03ce971 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1126,7 +1126,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; - channels = (hdr->bLength - 6) / 4; + channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; } -- 1.7.1

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[PATCH] ALSA: usb-audio: fix feature unit parser for UAC2
by Daniel Mack 27 May '10

27 May '10

Fix a small off-by-one bug which causes the feature unit to announce a wrong number of channels. This leads to illegal requests sent to the firmware eventually. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/mixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 97dd176..03ce971 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1126,7 +1126,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; - channels = (hdr->bLength - 6) / 4; + channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; } -- 1.7.1

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[PATCH] ALSA: usb-audio: parse clock topology of UAC2 devices
by Daniel Mack 27 May '10

27 May '10

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 <daniel(a)caiaq.de> --- 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 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 <linux/bitops.h> +#include <linux/init.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/string.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> + +#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 */ }; /* -- 1.7.1

1 0

Re: [alsa-devel] No mixers on ATI RS780 Azalia
by Clemens Ladisch 27 May '10

27 May '10

Robert Hancock wrote: > On 05/27/2010 06:34 AM, Jan Engelhardt wrote: >> On Thursday 2010-05-27 13:42, Clemens Ladisch wrote: >>> It is possible that an embedded device like this requires some custom >>> initialization. Can you find out if the Windows driver is the standard >>> Microsoft driver or Samsung's? >> >> Looks like the Microsoft one: > > Well, not Samsung, but it's not the Microsoft HDA driver either, it's > using a Realtek driver. > >> == "ATI HDMI Audio" == >> ... >> C:\WINDOWS\system32\drivers\RtHDMI.sys >> Realtek, 5.10.0.5602 built by WinDDK Googling shows that this is the driver that AMD ships for its "ATI HDMI" devices. >> == "Realtek High Definition Audio" == >> ... This also looks like the standard driver. Apparently, this computer uses standard audio hardware; the culprit is probably the Linux radeon driver. Regards, Clemens

1 0

[alsa-devel] [PATCH] imx-ssi.c: fix I2S slave setup
by Eric Bénard 27 May '10

27 May '10

* without this fix, the waveform on the SSI port are not real I2S signal (at least on i.MX27's SSI4) * original Freescale's driver has if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) && ssi_mode->network_mode) { scr &= ~SSI_SCR_I2S_MODE_MASK; scr |= SSI_SCR_I2S_MODE_SLAVE; } So as now the SSI is set in network mode as a default when running in I2S format, it seems reasonable to also set I2S_MODE_SLAVE Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/imx-ssi.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 80b4fee..c5e1626 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, /* * SSI DAI format configuration. * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S because the I2S mode is only a register preset. */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -98,7 +96,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) case SND_SOC_DAIFMT_I2S: /* data on rising edge of bclk, frame low 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; - scr |= SSI_SCR_NET; + scr |= SSI_SCR_NET | SSI_SCR_I2S_MODE_SLAVE; break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ -- 1.6.3.3

3 4

[alsa-devel] [PATCH] alsa-python: add register_poll to alsaseq
by Hector Martin 27 May '10

27 May '10

Attached patch adds register_poll to the sequencer bindings to support nonblocking sequencer access with the select module. I sent this last month but it looks like it was overlooked. Resending. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

2 1

[alsa-devel] [PATCH: alsa-lib] Fix a string format ambiguity
by Colin Guthrie 27 May '10

27 May '10

Hi, Simply patch to fix a format-string ambiguity. Meant to give this over before 1.0.23 but forgot. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

2 1

[alsa-devel] [PATCH] Add support for Thinkpad Edge conexant chip
by Jerone Young 27 May '10

27 May '10

This quirks in support for the Thinkpad Edge. Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index e863649..f4a2bd6 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -2975,6 +2975,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),

2 1

[alsa-devel] [PATCH 1/5] Remove unused io map functions.
by linux＠audioscience.com 27 May '10

27 May '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpios.c | 23 ----------------------- sound/pci/asihpi/hpios.h | 9 --------- 2 files changed, 0 insertions(+), 32 deletions(-) diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index de615cf..742ee12 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -89,26 +89,3 @@ u16 hpios_locked_mem_free(struct consistent_dma_area *p_mem_area) void hpios_locked_mem_free_all(void) { } - -void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx, - unsigned int length) -{ - HPI_DEBUG_LOG(DEBUG, "mapping %d %s %08llx-%08llx %04llx len 0x%x\n", - idx, pci_dev->resource[idx].name, - (unsigned long long)pci_resource_start(pci_dev, idx), - (unsigned long long)pci_resource_end(pci_dev, idx), - (unsigned long long)pci_resource_flags(pci_dev, idx), length); - - if (!(pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM)) { - HPI_DEBUG_LOG(ERROR, "not an io memory resource\n"); - return NULL; - } - - if (length > pci_resource_len(pci_dev, idx)) { - HPI_DEBUG_LOG(ERROR, "resource too small for requested %d \n", - length); - return NULL; - } - - return ioremap(pci_resource_start(pci_dev, idx), length); -} diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index a62c3f1..370f39b 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -166,13 +166,4 @@ struct hpi_adapter { void __iomem *ap_remapped_mem_base[HPI_MAX_ADAPTER_MEM_SPACES]; }; -static inline void hpios_unmap_io(void __iomem *addr, - unsigned long size) -{ - iounmap(addr); -} - -void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx, - unsigned int length); - #endif -- 1.7.0.4

2 5

[alsa-devel] [PATCH 1/4] ALSA: usb-audio: parse more format descriptors with structs
by Daniel Mack 27 May '10

27 May '10

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/endpoint.c | 11 +++++++---- sound/usb/format.c | 20 ++++++++++---------- sound/usb/format.h | 7 ++++--- 3 files changed, 21 insertions(+), 17 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index ef07a6d..4887342 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -158,8 +158,9 @@ 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; - unsigned char *fmt, *csep; + unsigned char *csep; int num, protocol; + struct uac_format_type_i_continuous_descriptor *fmt; dev = chip->dev; @@ -256,8 +257,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno); continue; } - if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || - ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || + ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; @@ -268,7 +269,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * with the previous one, except for a larger packet size, but * is actually a mislabeled two-channel setting; ignore it. */ - if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && + if (fmt->bNrChannels == 1 && + fmt->bSubframeSize == 2 && + altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->formats == SNDRV_PCM_FMTBIT_S16_LE && protocol == UAC_VERSION_1 && diff --git a/sound/usb/format.c b/sound/usb/format.c index b87cf87..caaa3ef 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -278,12 +278,11 @@ err: * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, - struct audioformat *fp, - int format, void *_fmt, + struct audioformat *fp, int format, + struct uac_format_type_i_continuous_descriptor *fmt, struct usb_host_interface *iface) { struct usb_interface_descriptor *altsd = get_iface_desc(iface); - struct uac_format_type_i_discrete_descriptor *fmt = _fmt; int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; @@ -320,7 +319,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, switch (protocol) { case UAC_VERSION_1: fp->channels = fmt->bNrChannels; - ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); break; case UAC_VERSION_2: /* fp->channels is already set in this case */ @@ -392,12 +391,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, } int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface) + int format, struct uac_format_type_i_continuous_descriptor *fmt, + int stream, struct usb_host_interface *iface) { int err; - switch (fmt[3]) { + switch (fmt->bFormatType) { case UAC_FORMAT_TYPE_I: case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); @@ -407,10 +406,11 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, + fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -421,7 +421,7 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == UAC_FORMAT_TYPE_I && + if (fmt->bFormatType == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; diff --git a/sound/usb/format.h b/sound/usb/format.h index 8298c4e..387924f 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -1,8 +1,9 @@ #ifndef __USBAUDIO_FORMAT_H #define __USBAUDIO_FORMAT_H -int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface); +int snd_usb_parse_audio_format(struct snd_usb_audio *chip, + struct audioformat *fp, int format, + struct uac_format_type_i_continuous_descriptor *fmt, + int stream, struct usb_host_interface *iface); #endif /* __USBAUDIO_FORMAT_H */ -- 1.7.1

2 5

Re: [alsa-devel] [PATCH 17/17] sound: Add missing spin_unlock
by Takashi Iwai 27 May '10

27 May '10

At Wed, 26 May 2010 17:59:27 +0200 (CEST), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > Add a spin_unlock missing on the error path. > > The semantic match that finds this problem is as follows: > (http://coccinelle.lip6.fr/) > > // <smpl> > @@ > expression E1; > @@ > > * spin_lock(E1,...); > <+... when != E1 > if (...) { > ... when != E1 > * return ...; > } > ...+> > * spin_unlock(E1,...); > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Thanks, applied now. Takashi > --- > sound/mips/au1x00.c | 1 + > sound/oss/dmasound/dmasound_atari.c | 5 +++-- > 2 files changed, 4 insertions(+), 2 deletions(-) > > diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c > index 3e763d6..446cf97 100644 > --- a/sound/mips/au1x00.c > +++ b/sound/mips/au1x00.c > @@ -516,6 +516,7 @@ get the interrupt driven case to work efficiently */ > break; > if (i == 0x5000) { > printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); > + spin_unlock(&au1000->ac97_lock); > return 0; > } > > diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c > index 1f47741..13c2144 100644 > --- a/sound/oss/dmasound/dmasound_atari.c > +++ b/sound/oss/dmasound/dmasound_atari.c > @@ -1277,7 +1277,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > * (almost) like on the TT. > */ > write_sq_ignore_int = 0; > - return IRQ_HANDLED; > + goto out; > } > > if (!write_sq.active) { > @@ -1285,7 +1285,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > * the sq variables, so better don't do anything here. > */ > WAKE_UP(write_sq.sync_queue); > - return IRQ_HANDLED; > + goto out; > } > > /* Probably ;) one frame is finished. Well, in fact it may be that a > @@ -1322,6 +1322,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > /* We are not playing after AtaPlay(), so there > is nothing to play any more. Wake up a process > waiting for audio output to drain. */ > +out: > spin_unlock(&dmasound.lock); > return IRQ_HANDLED; > } >

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 27 May '10

27 May '10

The following changes since commit ad8332c1302bcb4f80d593fd3eb477be9d7f5604: ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies (2010-05-20 10:28:39 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 This fixes up the build of the i.MX SSI driver, a minor warning fix for SH and fix the dB scales for a bunch of Wolfson CODECs. Guennadi Liakhovetski (1): ASoC: fix uninitialised variable in siu_dai.c Mark Brown (3): ASoC: Fix dB scales for WM835x ASoC: Fix dB scales for WM8400 ASoC: Fix dB scales for WM8990 Stuart Longland (1): ASoC: Update Freescale i.MX SSI driver DMA parameter handling sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8400.c | 18 +++++++++--------- sound/soc/codecs/wm8990.c | 18 +++++++++--------- sound/soc/imx/imx-pcm-dma-mx2.c | 7 ++++--- sound/soc/sh/siu_dai.c | 2 ++ 5 files changed, 26 insertions(+), 23 deletions(-)

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[alsa-devel] [PATCH 1/2] ASOC: S5PC100: Enable AC97 support
by Jassi Brar 27 May '10

27 May '10

From: Jassi Brar <jassi.brar(a)samsung.com> The S5PC100 has the AC97 controller same as S3C6410. Simply enable the options to build the drivers for S5PC100 also. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/smdk_wm9713.c | 1 + 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..aa112d3 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to @@ -120,7 +120,7 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 24fd39f..7dd85e5 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -25,6 +25,7 @@ static struct snd_soc_card smdk; * Default CFG switch settings to use this driver: * * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + * SMDKC100: Set CFG6 1-3 On, CFG7 1 On */ /* -- 1.6.2.5

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[alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by VDR User 27 May '10

27 May '10

Hi. I had been mostly successfully using the snd_hda_intel driver from the stable kernel 2.6.34 for HDMI audio with my Nvidia GT220 video card. In an attempt to see if any recent changes to snd_hda_intel would fix a problem I've been having with "pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed" I decided to download the latest snapshot. Unfortunately with these drivers I have no audio. All my settings, and everything have remained the same. The only noticable difference I've found is /sys/module/snd_hda_intel/parameters/probe_only is 0,0,0,0,0,0,0,0 in the snapshot drivers, and N,N,N,N,N,N,N,N in the 2.6.34 kernel drivers (which are compiled into the kernel). Any help? Thanks, Derek

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[alsa-devel] [PATCH 2/2] ASOC: S5PV210: Enable AC97 support
by Jassi Brar 27 May '10

27 May '10

From: Jassi Brar <jassi.brar(a)samsung.com> The S5PV210 and S5PC110 has the AC97 controller same as S3C6410. Simply enable the options to build the drivers for S5PC110 and S5PV210 also. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/smdk_wm9713.c | 2 ++ 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index aa112d3..292d817 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to @@ -120,7 +120,7 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100) + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 7dd85e5..5527b9e 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -26,6 +26,8 @@ static struct snd_soc_card smdk; * * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off * SMDKC100: Set CFG6 1-3 On, CFG7 1 On + * SMDKC110: Set CFGB10 1-2 Off, CFGB12 1-3 On + * SMDKV210: Set CFGB10 1-2 Off, CFGB12 1-3 On */ /* -- 1.6.2.5

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[alsa-devel] [RFC PATCH v2 0/3] ep93xx i2s audio
by Ryan Mallon 27 May '10

27 May '10

The following patch series adds support for i2s audio on the ep93xx, with the Snapper CL15 used as an example implementation. I have created psuedo clocks for sclk and lrclk as suggested by Hartley/Mark. This moves the sdiv/lrdiv calculation into ep93xx_i2s_hw_parms. The ep93xx documentation implies that lrclk is gated for unused bits (ie if lrdiv > word length). If this is true for all ep93xx i2s setups then the lrclk/sclk's could be replaced by a single audio rate clock. I have fixed the enabling/disabling of the i2s peripheral, which corrects the problem I was having with playback sometimes resulting in noise. I have also made a number of changes suggested from the first round of patches. Ryan Mallon (3): ep93xx i2s audio driver ep93xx i2s core support ep93xx i2s audio snapper cl15 support arch/arm/mach-ep93xx/clock.c | 69 ++++- arch/arm/mach-ep93xx/core.c | 31 ++ arch/arm/mach-ep93xx/include/mach/ep93xx-regs.h | 10 + arch/arm/mach-ep93xx/include/mach/platform.h | 1 + arch/arm/mach-ep93xx/snappercl15.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/ep93xx/Kconfig | 18 + sound/soc/ep93xx/Makefile | 11 + sound/soc/ep93xx/ep93xx-i2s.c | 489 +++++++++++++++++++++++ sound/soc/ep93xx/ep93xx-i2s.h | 25 ++ sound/soc/ep93xx/ep93xx-pcm.c | 323 +++++++++++++++ sound/soc/ep93xx/ep93xx-pcm.h | 22 + sound/soc/ep93xx/snappercl15.c | 143 +++++++ 14 files changed, 1144 insertions(+), 1 deletions(-) create mode 100644 sound/soc/ep93xx/Kconfig create mode 100644 sound/soc/ep93xx/Makefile create mode 100644 sound/soc/ep93xx/ep93xx-i2s.c create mode 100644 sound/soc/ep93xx/ep93xx-i2s.h create mode 100644 sound/soc/ep93xx/ep93xx-pcm.c create mode 100644 sound/soc/ep93xx/ep93xx-pcm.h create mode 100644 sound/soc/ep93xx/snappercl15.c

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[alsa-devel] [patch 0/5] kirkwood openrd client audio support - v3
by apatard＠mandriva.com 27 May '10

27 May '10

Hi, It's a new try to get merged kirkwood openrd client support. The changes are : - set window configuration according to dma address - namespaced all cs42l51 #define - modified cs42l51_dac_names array according to what proposed Mark Brown - use SOC_DOUBLE_TLV for mic boost Arnaud

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[PATCH 3/4] eukrea_mbimx27: add audio codec
by Eric Bénard 26 May '10

26 May '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c | 44 +++++++++++++++++++++++++- 1 files changed, 43 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c b/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c index 34bdc26..471dbd2 100644 --- a/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c +++ b/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2009 Eric Benard - eric(a)eukrea.com + * Copyright (C) 2009-2010 Eric Benard - eric(a)eukrea.com * * Based on pcm970-baseboard.c which is : * Copyright (C) 2008 Juergen Beisert (kernel(a)pengutronix.de) @@ -37,6 +37,8 @@ #include <mach/mmc.h> #include <mach/imx-uart.h> #include <mach/spi.h> +#include <mach/ssi.h> +#include <mach/audmux.h> #include "devices.h" @@ -93,6 +95,13 @@ static int eukrea_mbimx27_pins[] = { PD29_PF_CSPI1_SCLK, PD30_PF_CSPI1_MISO, PD31_PF_CSPI1_MOSI, + /* SSI4 */ +#if defined(CONFIG_SND_SOC_EUKREA_TLV320) + PC16_PF_SSI4_FS, + PC17_PF_SSI4_RXD | GPIO_PUEN, + PC18_PF_SSI4_TXD | GPIO_PUEN, + PC19_PF_SSI4_CLK, +#endif }; static const uint32_t eukrea_mbimx27_keymap[] = { @@ -287,6 +296,12 @@ static struct spi_board_info eukrea_mbimx27_spi_board_info[] __initdata = { }, }; +static struct i2c_board_info eukrea_mbimx27_i2c_devices[] = { + { + I2C_BOARD_INFO("tlv320aic23", 0x1a), + }, +}; + static int eukrea_mbimx27_spi_cs[] = {GPIO_PORTD | 28}; static struct spi_imx_master eukrea_mbimx27_spi_0_data = { @@ -303,6 +318,10 @@ static struct imxmmc_platform_data sdhc_pdata = { .dat3_card_detect = 1, }; +struct imx_ssi_platform_data eukrea_mbimx27_ssi_pdata = { + .flags = IMX_SSI_DMA, +}; + /* * system init for baseboard usage. Will be called by cpuimx27 init. * @@ -314,6 +333,24 @@ void __init eukrea_mbimx27_baseboard_init(void) mxc_gpio_setup_multiple_pins(eukrea_mbimx27_pins, ARRAY_SIZE(eukrea_mbimx27_pins), "MBIMX27"); +#if defined(CONFIG_SND_SOC_EUKREA_TLV320) + /* SSI unit master I2S codec connected to SSI_PINS_4*/ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_TCLKDIR | + MXC_AUDMUX_V1_PCR_RFSDIR | + MXC_AUDMUX_V1_PCR_RCLKDIR | + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); +#endif + mxc_register_device(&mxc_uart_device1, &uart_pdata[0]); mxc_register_device(&mxc_uart_device2, &uart_pdata[1]); #if !defined(MACH_EUKREA_CPUIMX27_USEUART4) @@ -323,6 +360,11 @@ void __init eukrea_mbimx27_baseboard_init(void) mxc_register_device(&mxc_fb_device, &eukrea_mbimx27_fb_data); mxc_register_device(&mxc_sdhc_device0, &sdhc_pdata); + i2c_register_board_info(0, eukrea_mbimx27_i2c_devices, + ARRAY_SIZE(eukrea_mbimx27_i2c_devices)); + + mxc_register_device(&imx_ssi_device0, &eukrea_mbimx27_ssi_pdata); + #if defined(CONFIG_TOUCHSCREEN_ADS7846) \ || defined(CONFIG_TOUCHSCREEN_ADS7846_MODULE) /* ADS7846 Touchscreen controller init */ -- 1.6.3.3 --------------010207020109090402010001--

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[PATCH 2/4] soc/imx: add eukrea-tlv320
by Eric Bénard 26 May '10

26 May '10

Add the necessary files to support the TLV320AIC23B wired in I2S on our i.MX platforms. Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 8 +++ sound/soc/imx/Makefile | 2 + sound/soc/imx/eukrea-tlv320.c | 131 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 141 insertions(+), 0 deletions(-) create mode 100644 sound/soc/imx/eukrea-tlv320.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 7174b4c..3e27b16 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -11,3 +11,11 @@ config SND_IMX_SOC config SND_MXC_SOC_SSI tristate +config SND_SOC_EUKREA_TLV320 + bool "Eukrea TLV320" + depends on MACH_EUKREA_MBIMX27_BASEBOARD + select SND_IMX_SOC + select SND_SOC_TLV320AIC23 + help + Enable I2S based access to the TLV320AIC23B codec attached + to the SSI4 interface diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 9f8bb92..28d4b1e 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -10,3 +10,5 @@ obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support snd-soc-phycore-ac97-objs := phycore-ac97.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o +obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c new file mode 100644 index 0000000..e368787 --- /dev/null +++ b/sound/soc/imx/eukrea-tlv320.c @@ -0,0 +1,131 @@ +/* + * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode + * + * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric(a)eukrea.com> + * + * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * which is Copyright 2009 Simtec Electronics + * and on sound/soc/imx/phycore-ac97.c which is + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de> + * + * 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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#include "../codecs/tlv320aic23.h" +#include "imx-ssi.h" + +#define CODEC_CLOCK 12000000 + +static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops eukrea_tlv320_snd_ops = { + .hw_params = eukrea_tlv320_hw_params, +}; + +static struct snd_soc_dai_link eukrea_tlv320_dai = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .codec_dai = &tlv320aic23_dai, + .ops = &eukrea_tlv320_snd_ops, +}; + +static struct snd_soc_card eukrea_tlv320 = { + .name = "cpuimx-audio", + .platform = &imx_soc_platform, + .dai_link = &eukrea_tlv320_dai, + .num_links = 1, +}; + +static struct snd_soc_device eukrea_tlv320_snd_devdata = { + .card = &eukrea_tlv320, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *eukrea_tlv320_snd_device; + +static int __init eukrea_tlv320_init(void) +{ + int ret; + + if (!machine_is_eukrea_cpuimx27()) + /* return happy. We might run on a totally different machine */ + return 0; + + eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); + if (!eukrea_tlv320_snd_device) + return -ENOMEM; + + eukrea_tlv320_dai.cpu_dai = &imx_ssi_pcm_dai[0]; + + platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320_snd_devdata); + eukrea_tlv320_snd_devdata.dev = &eukrea_tlv320_snd_device->dev; + ret = platform_device_add(eukrea_tlv320_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(eukrea_tlv320_snd_device); + } + + return ret; +} + +static void __exit eukrea_tlv320_exit(void) +{ + platform_device_unregister(eukrea_tlv320_snd_device); +} + +module_init(eukrea_tlv320_init); +module_exit(eukrea_tlv320_exit); + +MODULE_AUTHOR("Eric Bénard <eric(a)eukrea.com>"); +MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- 1.6.3.3 --------------010207020109090402010001 Content-Type: text/x-patch; name="0003-eukrea_mbimx27-add-audio-codec.patch" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename="0003-eukrea_mbimx27-add-audio-codec.patch"

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Re: [alsa-devel] CX20582 (Pebble) and Thinkpad x100e...
by Tobin Davis 26 May '10

26 May '10

I'm forwarding this on to the alsa-devel mailing list as I am not personally maintaining this driver at this time. If you could also download alsa-info.sh from http://alsa-project.org, run it, and post the results to the mailing list that would be great. -- Tobin Davis "Look! There! Evil!.. pure and simple, total evil from the Eighth Dimension!" -- Buckaroo Banzai On Mon, 2010-05-24 at 22:20 +0300, Hakan Kızılhan wrote: > Pototskiy Akex > Takashi Iwai > Tobin Davis > > Hi, > > I am Hakan from Turkey. If no drawback, i want to ask for a hardware problem. > > I have AMD based Thinkpad x100e and this machine have Conexant CX20582 > (Pebble) sound card. > Using Ubuntu 10.04 and microphones doesn't work. I have readed following bug: > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/549289 > > options snd-hda-intel model="olpc-xo-1_5" enables output > jack(headphones) but microphones not. > i dont capture sound anyway. > > i saw "sound/pci/hda/patch_conexant.c" related patches for different models. > > So, is it posible to patch this file for Thinkpad x100e model? > May i send hardware specific information for patch? And what should i send? > > Thanks a lot... > > ____________ > Hakan Kizilhan

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Re: [alsa-devel] [patch 4/5] orion/kirkwood: Add i2s support
by Arnaud Patard 26 May '10

26 May '10

Nicolas Pitre <nico(a)fluxnic.net> writes: Hi, > On Tue, 25 May 2010, apatard(a)mandriva.com wrote: > >> This patch enables support for the i2s controller available on orion/kirkwood platforms >> >> Signed-off-by: Arnaud Patard <apatard(a)mandriva.com> > > The actual Orion processor family has no I2S/audio support. It would be > better if all the audio related stuff was renamed to "kirkwood" and > "KIRKWOOD_*" instead. I've been told there was some orion with audio. As I've never looked at the orion family in details, I assumed this is right and I've changed the naming accordingly. I can change again but it would be nice to have an agreement here. Who's/what's right ? orion can or can not have i2s audio ? Arnaud

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