18 files changed, 59 insertions, 35 deletions

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/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/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/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..18c7c91786bc 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -26,7 +26,7 @@
  * SPI 0 -> 1st PCM1796 (front)
  * SPI 1 -> 2nd PCM1796 (surround)
  * SPI 2 -> 3rd PCM1796 (center/LFE)
- * SPI 4 -> 4th PCM1796 (back)
+ * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!)
  *
  * GPIO 2 -> M0 of CS5381
  * GPIO 3 -> M1 of CS5381
@@ -207,6 +207,12 @@ static void xonar_gpio_changed(struct oxygen *chip);
 static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
                                      u8 reg, u8 value)
 {
+        /*
+         * We don't want to do writes on SPI 4 because the EEPROM, which shares
+         * the same pin, might get confused and broken.  We'd better take care
+         * that the driver works with the default register values ...
+         */
+#if 0
         /* maps ALSA channel pair number to SPI output */
         static const u8 codec_map[4] = {
                 0, 1, 2, 4
@@ -217,6 +223,7 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
                          (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
                          OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
                          (reg << 8) | value);
+#endif
 }
 
 static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec,
@@ -676,7 +683,7 @@ 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;
@@ -750,6 +757,9 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
 
 static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 {
+        if (!strncmp(template->name, "Master Playback ", 16))
+                /* disable volume/mute because they would require SPI writes */
+                return 1;
         if (!strncmp(template->name, "CD Capture ", 11))
                 /* CD in is actually connected to the video in pin */
                 template->private_value ^= AC97_CD ^ AC97_VIDEO;
@@ -840,9 +850,8 @@ static const struct oxygen_model model_xonar_d2 = {
         .dac_volume_min = 0x0f,
         .dac_volume_max = 0xff,
         .misc_flags = OXYGEN_MISC_MIDI,
-        .function_flags = OXYGEN_FUNCTION_SPI |
-                          OXYGEN_FUNCTION_ENABLE_SPI_4_5,
-        .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+        .function_flags = OXYGEN_FUNCTION_SPI,
+        .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S,
         .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
@@ -899,6 +908,7 @@ 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/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/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..1cbb7b9b51ce 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
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/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..2ab83129d9b0 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2966,6 +2966,7 @@ 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/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