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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

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* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  sound: do not set DEVNAME for OSS devices
  ALSA: hda - Add sanity check in PCM open callback
  ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback
  ALSA: hda - Avoid invalid formats and rates with shared SPDIF
  ALSA: hda - Improve ASUS eeePC 1000 mixer
  ALSA: hda - Add GPIO1 control at muting with HP laptops
  ALSA: usx2y - reparent sound device
  ALSA: snd_usb_caiaq: reparent sound device
  sound: virtuoso: fix Xonar D1/DX silence after resume
  ASoC: Only disable pxa2xx-i2s clocks if we enabled them
  ALSA: hda - Add quirk for HP 6930p
  ALSA: hda - Add missing static to patch_ca0110()
  ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
  ASoC: remove BROKEN from Efika and pcm030 fabric drivers
  ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig
This commit is contained in:
Linus Torvalds 2009-07-04 09:46:13 -07:00
commit f1a745710f
13 changed files with 80 additions and 40 deletions
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14
sound/pci/hda/hda_codec.c
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@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
}
mutex_lock(&codec->spdif_mutex);
if (mout->share_spdif) {
runtime->hw.rates &= mout->spdif_rates;
runtime->hw.formats &= mout->spdif_formats;
if (mout->spdif_maxbps < hinfo->maxbps)
hinfo->maxbps = mout->spdif_maxbps;
if ((runtime->hw.rates & mout->spdif_rates) &&
(runtime->hw.formats & mout->spdif_formats)) {
runtime->hw.rates &= mout->spdif_rates;
runtime->hw.formats &= mout->spdif_formats;
if (mout->spdif_maxbps < hinfo->maxbps)
hinfo->maxbps = mout->spdif_maxbps;
} else {
mout->share_spdif = 0;
/* FIXME: need notify? */
}
}
mutex_unlock(&codec->spdif_mutex);
}

7
sound/pci/hda/hda_intel.c
View File

@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
mutex_unlock(&chip->open_mutex);
return err;
}
snd_pcm_limit_hw_rates(runtime);
spin_lock_irqsave(&chip->reg_lock, flags);
azx_dev->substream = substream;
azx_dev->running = 0;
@ -1463,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
snd_pcm_set_sync(substream);
mutex_unlock(&chip->open_mutex);
if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max))
return -EINVAL;
if (snd_BUG_ON(!runtime->hw.formats))
return -EINVAL;
if (snd_BUG_ON(!runtime->hw.rates))
return -EINVAL;
return 0;
}

28
sound/pci/hda/patch_analog.c
View File

@ -3746,9 +3746,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
{ } /* end */
};
static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
int mute = (!ucontrol->value.integer.value[0] &&
!ucontrol->value.integer.value[1]);
/* toggle GPIO1 according to the mute state */
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
mute ? 0x02 : 0x0);
return ret;
}
static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = ad1884a_mobile_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
@ -3869,6 +3890,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = {
/* unsolicited event for pin-sense */
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
/* allow to touch GPIO1 (for mute control) */
{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
{ } /* end */
};
@ -3978,6 +4003,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),

2
sound/pci/hda/patch_ca0110.c
View File

@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec)
}
int patch_ca0110(struct hda_codec *codec)
static int patch_ca0110(struct hda_codec *codec)
{
struct ca0110_spec *spec;
int err;

24
sound/pci/hda/patch_realtek.c
View File

@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
{ }
};
/* bind volumes of both NID 0x0c and 0x0d */
static struct hda_bind_ctls alc269_epc_bind_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
0
},
};
static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
};
/* FSC amilo */
static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol),
{ } /* end */
};
#define alc269_fujitsu_mixer alc269_eeepc_mixer
static struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},

2
sound/pci/oxygen/virtuoso.c
View File

@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip)
static void xonar_d1_resume(struct oxygen *chip)
{
oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
msleep(1);
cs43xx_init(chip);
xonar_enable_output(chip);
}

6
sound/soc/fsl/Kconfig
View File

@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S
config SND_SOC_MPC5200_AC97
tristate "Freescale MPC5200 PSC in AC97 mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
select AC97_BUS
select SND_SOC_AC97_BUS
select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
@ -41,7 +41,7 @@ config SND_SOC_MPC5200_AC97
config SND_MPC52xx_SOC_PCM030
tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
depends on PPC_MPC5200_SIMPLE && BROKEN
depends on PPC_MPC5200_SIMPLE
select SND_SOC_MPC5200_AC97
select SND_SOC_WM9712
help
@ -50,7 +50,7 @@ config SND_MPC52xx_SOC_PCM030
config SND_MPC52xx_SOC_EFIKA
tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
depends on PPC_EFIKA && BROKEN
depends on PPC_EFIKA
select SND_SOC_MPC5200_AC97
select SND_SOC_STAC9766
help

11
sound/soc/omap/omap-pcm.c
View File

@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
dma_addr_t ptr;
snd_pcm_uframes_t offset;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ptr = omap_get_dma_src_pos(prtd->dma_ch);
else
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ptr = omap_get_dma_dst_pos(prtd->dma_ch);
offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
} else if (!(cpu_is_omap1510())) {
ptr = omap_get_dma_src_pos(prtd->dma_ch);
offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
} else
offset = prtd->period_index * runtime->period_size;
offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
if (offset >= runtime->buffer_size)
offset = 0;

7
sound/soc/pxa/pxa2xx-i2s.c
View File

@ -167,6 +167,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
BUG_ON(IS_ERR(clk_i2s));
clk_enable(clk_i2s);
dai->private_data = dai;
pxa_i2s_wait();
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@ -255,7 +256,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
SACR0 &= ~SACR0_ENB;
pxa_i2s_wait();
clk_disable(clk_i2s);
if (dai->private_data != NULL) {
clk_disable(clk_i2s);
dai->private_data = NULL;
}
}
}
@ -336,6 +340,7 @@ static int pxa2xx_i2s_probe(struct platform_device *dev)
return PTR_ERR(clk_i2s);
pxa_i2s_dai.dev = &dev->dev;
pxa_i2s_dai.private_data = NULL;
ret = snd_soc_register_dai(&pxa_i2s_dai);
if (ret != 0)
clk_put(clk_i2s);

5
sound/sound_core.c
View File

@ -10,6 +10,8 @@
#include <linux/module.h>
#include <linux/device.h>
#include <linux/err.h>
#include <linux/kdev_t.h>
#include <linux/major.h>
#include <sound/core.h>
#ifdef CONFIG_SOUND_OSS_CORE
@ -29,6 +31,8 @@ MODULE_LICENSE("GPL");
static char *sound_nodename(struct device *dev)
{
if (MAJOR(dev->devt) == SOUND_MAJOR)
return NULL;
return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev));
}
@ -104,7 +108,6 @@ module_exit(cleanup_soundcore);
#include <linux/types.h>
#include <linux/kernel.h>
#include <linux/sound.h>
#include <linux/major.h>
#include <linux/kmod.h>
#define SOUND_STEP 16

10
sound/usb/caiaq/device.c
View File

@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17");
MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
log("Unable to set up control system (ret=%d)\n", ret);
}
static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
static int create_card(struct usb_device *usb_dev,
struct usb_interface *intf,
struct snd_card **cardp)
{
int devnum;
int err;
@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor),
le16_to_cpu(usb_dev->descriptor.idProduct));
spin_lock_init(&dev->spinlock);
snd_card_set_dev(card, &usb_dev->dev);
snd_card_set_dev(card, &intf->dev);
*cardp = card;
return 0;
@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
struct snd_card *card;
struct usb_device *device = interface_to_usbdev(intf);
ret = create_card(device, &card);
ret = create_card(device, intf, &card);
if (ret < 0)
return ret;

2
sound/usb/usx2y/us122l.c
View File

@ -514,7 +514,6 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp)
US122L(card)->chip.dev->bus->busnum,
US122L(card)->chip.dev->devnum
);
snd_card_set_dev(card, &device->dev);
*cardp = card;
return 0;
}
@ -531,6 +530,7 @@ static int us122l_usb_probe(struct usb_interface *intf,
if (err < 0)
return err;
snd_card_set_dev(card, &intf->dev);
if (!us122l_create_card(card)) {
snd_card_free(card);
return -EINVAL;

2
sound/usb/usx2y/usbusx2y.c
View File

@ -364,7 +364,6 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp)
0,//us428(card)->usbmidi.ifnum,
usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum
);
snd_card_set_dev(card, &device->dev);
*cardp = card;
return 0;
}
@ -388,6 +387,7 @@ static int usX2Y_usb_probe(struct usb_device *device,
err = usX2Y_create_card(device, &card);
if (err < 0)
return err;
snd_card_set_dev(card, &intf->dev);
if ((err = usX2Y_hwdep_new(card, device)) < 0 ||
(err = snd_card_register(card)) < 0) {
snd_card_free(card);