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-rw-r--r--sound/arm/aaci.c6
-rw-r--r--sound/core/jack.c2
-rw-r--r--sound/core/oss/pcm_oss.c2
-rw-r--r--sound/core/oss/rate.c2
-rw-r--r--sound/drivers/mtpav.c3
-rw-r--r--sound/oss/dmasound/dmasound_atari.c16
-rw-r--r--sound/pci/Kconfig4
-rw-r--r--sound/pci/aw2/aw2-alsa.c2
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c1
-rw-r--r--sound/pci/intel8x0.c2
-rw-r--r--sound/pci/oxygen/virtuoso.c3
-rw-r--r--sound/pci/pcxhr/pcxhr.h12
-rw-r--r--sound/ppc/snd_ps3.c4
-rw-r--r--sound/soc/atmel/atmel-pcm.c2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.h2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c11
-rw-r--r--sound/soc/codecs/wm8350.c2
-rw-r--r--sound/soc/codecs/wm8753.c9
-rw-r--r--sound/soc/codecs/wm8990.c7
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c18
-rw-r--r--sound/soc/omap/omap-mcbsp.c4
-rw-r--r--sound/soc/omap/omap-pcm.c5
-rw-r--r--sound/soc/omap/sdp3430.c4
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c3
-rw-r--r--sound/usb/usbaudio.c21
-rw-r--r--sound/usb/usbmidi.c1
-rw-r--r--sound/usb/usbquirks.h8
29 files changed, 95 insertions, 68 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 89096e811a4b..772901e41ecb 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
*/
do {
v = readl(aaci->base + AACI_SLFR);
- } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--);
+ } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
if (!timeout)
dev_err(&aaci->dev->dev,
@@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
*/
do {
v = readl(aaci->base + AACI_SLFR);
- } while ((v & SLFR_1TXB) && timeout--);
+ } while ((v & SLFR_1TXB) && --timeout);
if (!timeout) {
dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
@@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
do {
cond_resched();
v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
- } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--);
+ } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
if (!timeout) {
dev_err(&aaci->dev->dev, "timeout on RX valid\n");
diff --git a/sound/core/jack.c b/sound/core/jack.c
index dd4a12dc09aa..077a85262c1c 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device)
int err;
snprintf(jack->name, sizeof(jack->name), "%s %s",
- card->longname, jack->id);
+ card->shortname, jack->id);
jack->input_dev->name = jack->name;
/* Default to the sound card device. */
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index e17836680f49..0a1798eafb0b 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
AFMT_S8 | AFMT_U16_LE |
AFMT_U16_BE |
AFMT_S32_LE | AFMT_S32_BE |
- AFMT_S24_LE | AFMT_S24_LE |
+ AFMT_S24_LE | AFMT_S24_BE |
AFMT_S24_PACKED;
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index a466443c4a26..2fa9299a440d 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin,
while (dst_frames1 > 0) {
S1 = S2;
if (src_frames1-- > 0) {
- S1 = *src;
+ S2 = *src;
src += src_step;
}
if (pos & ~R_MASK) {
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5b89c0883d60..48b64e6b2670 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -706,7 +706,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
mtp_card->card = card;
mtp_card->irq = -1;
mtp_card->share_irq = 0;
- mtp_card->inmidiport = 0xffffffff;
mtp_card->inmidistate = 0;
mtp_card->outmidihwport = 0xffffffff;
init_timer(&mtp_card->timer);
@@ -719,6 +718,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
if (err < 0)
goto __error;
+ mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST;
+
err = snd_mtpav_get_ISA(mtp_card);
if (err < 0)
goto __error;
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 57d9f154c88b..38931f2f6967 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -847,23 +847,23 @@ static int __init AtaIrqInit(void)
of events. So all we need to keep the music playing is
to provide the sound hardware with new data upon
an interrupt from timer A. */
- mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */
- mfp.tim_dt_a = 1; /* Cause interrupt after first event. */
- mfp.tim_ct_a = 8; /* Turn on event counting. */
+ st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */
+ st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */
+ st_mfp.tim_ct_a = 8; /* Turn on event counting. */
/* Register interrupt handler. */
if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound",
AtaInterrupt))
return 0;
- mfp.int_en_a |= 0x20; /* Turn interrupt on. */
- mfp.int_mk_a |= 0x20;
+ st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */
+ st_mfp.int_mk_a |= 0x20;
return 1;
}
#ifdef MODULE
static void AtaIrqCleanUp(void)
{
- mfp.tim_ct_a = 0; /* stop timer */
- mfp.int_en_a &= ~0x20; /* turn interrupt off */
+ st_mfp.tim_ct_a = 0; /* stop timer */
+ st_mfp.int_en_a &= ~0x20; /* turn interrupt off */
free_irq(IRQ_MFP_TIMA, AtaInterrupt);
}
#endif /* MODULE */
@@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void)
is_falcon = 0;
} else
return -ENODEV;
- if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0)
+ if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0)
return dmasound_init();
else {
printk("DMA sound driver: Timer A interrupt already in use\n");
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 6e3a1848447c..82b9bddcdcd6 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -744,8 +744,8 @@ config SND_VIRTUOSO
select SND_OXYGEN_LIB
help
Say Y here to include support for sound cards based on the
- Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and
- HDAV1.3 (Deluxe).
+ Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
+ Support for the HDAV1.3 (Deluxe) is very experimental.
To compile this driver as a module, choose M here: the module
will be called snd-virtuoso.
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f8..c7c54e7748e9 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
static struct pci_device_id snd_aw2_ids[] = {
- {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID,
+ {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
0, 0, 0},
{0}
};
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d66..101a1c13a20d 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
.driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]",
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 19d3391e229f..e900cdc84849 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip)
int time = 100;
if (chip->buggy_semaphore)
return 0; /* just ignore ... */
- while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
+ while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
udelay(1);
if (! time && ! chip->in_ac97_init)
snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n");
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 98c6a8c65d81..6c870c12a177 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -676,7 +676,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip)
if (chip->uart_input_count >= 2 &&
chip->uart_input[chip->uart_input_count - 2] == 'O' &&
chip->uart_input[chip->uart_input_count - 1] == 'K') {
- printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:");
+ printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n");
print_hex_dump_bytes("", DUMP_PREFIX_OFFSET,
chip->uart_input, chip->uart_input_count);
chip->uart_input_count = 0;
@@ -899,6 +899,7 @@ static const struct oxygen_model model_xonar_hdav = {
.dac_channels = 8,
.dac_volume_min = 0x0f,
.dac_volume_max = 0xff,
+ .misc_flags = OXYGEN_MISC_MIDI,
.function_flags = OXYGEN_FUNCTION_2WIRE,
.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c92..69d87dee6995 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -97,12 +97,12 @@ struct pcxhr_mgr {
int capture_chips;
int fw_file_set;
int firmware_num;
- int is_hr_stereo:1;
- int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */
- int board_has_analog:1; /* if 0 the board is digital only */
- int board_has_mic:1; /* if 1 the board has microphone input */
- int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
- int mono_capture:1; /* if 1 the board does mono capture */
+ unsigned int is_hr_stereo:1;
+ unsigned int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */
+ unsigned int board_has_analog:1; /* if 0 the board is digital only */
+ unsigned int board_has_mic:1; /* if 1 the board has microphone input */
+ unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
+ unsigned int mono_capture:1; /* if 1 the board does mono capture */
struct snd_dma_buffer hostport;
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 8f9e3859c37c..ff321110ec02 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -477,7 +477,7 @@ static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream)
card->dma_start_bus_addr[SND_PS3_CH_R] =
runtime->dma_addr + (runtime->dma_bytes / 2);
- pr_debug("%s: vaddr=%p bus=%#lx\n", __func__,
+ pr_debug("%s: vaddr=%p bus=%#llx\n", __func__,
card->dma_start_vaddr[SND_PS3_CH_L],
card->dma_start_bus_addr[SND_PS3_CH_L]);
@@ -1030,7 +1030,7 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
pr_info("%s: nullbuffer alloc failed\n", __func__);
goto clean_preallocate;
}
- pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__,
+ pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__,
the_card.null_buffer_start_vaddr,
the_card.null_buffer_start_dma_addr);
/* set default sample rate/word width */
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 1fac5efd285b..3dcdc4e3cfa0 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -44,8 +44,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <mach/hardware.h>
-
#include "atmel-pcm.h"
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index c5d67900d666..ff0054b76502 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -10,7 +10,7 @@
* Based on at91-ssc.c by
* Frank Mandarino <fmandarino@endrelia.com>
* Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index a828746e8a2f..391135f9c6c1 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -10,7 +10,7 @@
* Based on at91-ssc.c by
* Frank Mandarino <fmandarino@endrelia.com>
* Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b47a749c5ea2..aea0cb72d80a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
- int reg = kcontrol->private_value & 0xff;
- int shift = (kcontrol->private_value >> 8) & 0x0f;
- int mask = (kcontrol->private_value >> 16) & 0xff;
- int invert = (kcontrol->private_value >> 24) & 0x01;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
unsigned short val, val_mask;
int ret;
struct snd_soc_dapm_path *path;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e3989d406f54..35d99750c383 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
*
- * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6c21b50c9375..77620ab98756 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1451,7 +1451,14 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
},
};
-struct snd_soc_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[] = {
+ {
+ .name = "WM8753 DAI 0",
+ },
+ {
+ .name = "WM8753 DAI 1",
+ },
+};
EXPORT_SYMBOL_GPL(wm8753_dai);
static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5b5afc144478..a5731faa150c 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -2,8 +2,7 @@
* wm8990.c -- WM8990 ALSA Soc Audio driver
*
* Copyright 2008 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
@@ -177,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int reg = kcontrol->private_value & 0xff;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int reg = mc->reg;
int ret;
u16 val;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index bcec3f60bad9..acf39a646b2f 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
};
/**
- * mpc8610_hpcd_machine: ASoC machine data
- */
-static struct snd_soc_card mpc8610_hpcd_machine = {
- .probe = mpc8610_hpcd_machine_probe,
- .remove = mpc8610_hpcd_machine_remove,
- .name = "MPC8610 HPCD",
- .num_links = 1,
-};
-
-/**
* mpc8610_hpcd_probe: OF probe function for the fabric driver
*
* This function gets called when an SSI node is found in the device tree.
@@ -455,7 +445,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */
machine_data->dai.ops = &mpc8610_hpcd_ops;
- mpc8610_hpcd_machine.dai_link = &machine_data->dai;
+ machine_data->machine.probe = mpc8610_hpcd_machine_probe;
+ machine_data->machine.remove = mpc8610_hpcd_machine_remove;
+ machine_data->machine.name = "MPC8610 HPCD";
+ machine_data->machine.num_links = 1;
+ machine_data->machine.dai_link = &machine_data->dai;
/* Allocate a new audio platform device structure */
sound_device = platform_device_alloc("soc-audio", -1);
@@ -465,7 +459,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
goto error;
}
- machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
+ machine_data->sound_devdata.card = &machine_data->machine;
machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
machine_data->machine.platform = &fsl_soc_platform;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index ec5e18a78758..05dd5abcddf4 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -302,6 +302,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
regs->spcr1 |= RINTM(3);
regs->rcr2 |= RFIG;
regs->xcr2 |= XFIG;
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ regs->xccr = DXENDLY(1) | XDMAEN;
+ regs->rccr = RFULL_CYCLE | RDMAEN;
+ }
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b0362dfd5b71..dd3bb2933762 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -175,9 +175,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct omap_runtime_data *prtd = runtime->private_data;
+ unsigned long flags;
int ret = 0;
- spin_lock_irq(&prtd->lock);
+ spin_lock_irqsave(&prtd->lock, flags);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
@@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
default:
ret = -EINVAL;
}
- spin_unlock_irq(&prtd->lock);
+ spin_unlock_irqrestore(&prtd->lock, flags);
return ret;
}
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index ad97836818b1..e226fa75669c 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -91,7 +91,7 @@ static struct snd_soc_dai_link sdp3430_dai = {
};
/* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+static struct snd_soc_card snd_soc_sdp3430 = {
.name = "SDP3430",
.platform = &omap_soc_platform,
.dai_link = &sdp3430_dai,
@@ -100,7 +100,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = {
/* Audio subsystem */
static struct snd_soc_device sdp3430_snd_devdata = {
- .machine = &snd_soc_machine_sdp3430,
+ .card = &snd_soc_sdp3430,
.codec_dev = &soc_codec_dev_twl4030,
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 55fdb4abb179..ec3f8bb4b51d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
- if (ac97) {
+ /* Only instantiate AC97 if not already done by the adaptor
+ * for the generic AC97 subsystem.
+ */
+ if (ac97 && strcmp(codec->name, "AC97") != 0) {
ret = soc_ac97_dev_register(codec);
if (ret < 0) {
printk(KERN_ERR "asoc: AC97 device register failed\n");
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 493a4e8aa273..a2f1da8b4646 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -720,7 +720,8 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
struct snd_soc_dapm_path *path;
int found = 0;
- if (widget->id != snd_soc_dapm_mux)
+ if (widget->id != snd_soc_dapm_mux &&
+ widget->id != snd_soc_dapm_value_mux)
return -ENODEV;
if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c709b9563226..19e37451c216 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
* build the rate table and bitmap flags
*/
int r, idx;
- unsigned int nonzero_rates = 0;
fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
if (fp->rate_table == NULL) {
@@ -2532,24 +2531,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
return -1;
}
- fp->nr_rates = nr_rates;
- fp->rate_min = fp->rate_max = combine_triple(&fmt[8]);
+ fp->nr_rates = 0;
+ fp->rate_min = fp->rate_max = 0;
for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
unsigned int rate = combine_triple(&fmt[idx]);
+ if (!rate)
+ continue;
/* C-Media CM6501 mislabels its 96 kHz altsetting */
if (rate == 48000 && nr_rates == 1 &&
- chip->usb_id == USB_ID(0x0d8c, 0x0201) &&
+ (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
+ chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
fp->altsetting == 5 && fp->maxpacksize == 392)
rate = 96000;
- fp->rate_table[r] = rate;
- nonzero_rates |= rate;
- if (rate < fp->rate_min)
+ fp->rate_table[fp->nr_rates] = rate;
+ if (!fp->rate_min || rate < fp->rate_min)
fp->rate_min = rate;
- else if (rate > fp->rate_max)
+ if (!fp->rate_max || rate > fp->rate_max)
fp->rate_max = rate;
fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+ fp->nr_rates++;
}
- if (!nonzero_rates) {
+ if (!fp->nr_rates) {
hwc_debug("All rates were zero. Skipping format!\n");
return -1;
}
@@ -2966,6 +2968,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
return -EINVAL;
}
alts = &iface->altsetting[fp->altset_idx];
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
usb_set_interface(chip->dev, fp->iface, 0);
init_usb_pitch(chip->dev, fp->iface, alts, fp);
init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max);
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 320641ab5be7..26bad373fe65 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
}
ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.out_interval = 0;
ep_info.out_cables = endpoint->out_cables & 0x5555;
err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
if (err < 0)
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 92115755d98e..5d8ef09b9dcc 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -128,6 +128,14 @@
.bInterfaceClass = USB_CLASS_AUDIO,
.bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
},
+{
+ USB_DEVICE(0x046d, 0x0990),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Logitech, Inc.",
+ .product_name = "QuickCam Pro 9000",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
/*
* Yamaha devices
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-13 13:37:49 +0000