18 files changed, 66 insertions, 81 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 5c48e36038f2..dc78272fc39f 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -1089,7 +1089,7 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
                 goto out;
         }
 
-        aaci->base = ioremap(dev->res.start, SZ_4K);
+        aaci->base = ioremap(dev->res.start, resource_size(&dev->res));
         if (!aaci->base) {
                 ret = -ENOMEM;
                 goto out;
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index cdef2664218f..174dd2ff0f22 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -10,6 +10,7 @@
 #define __PCSP_H__
 
 #include <linux/hrtimer.h>
+#include <linux/timex.h>
 #if defined(CONFIG_MIPS) || defined(CONFIG_X86)
 /* Use the global PIT lock ! */
 #include <asm/i8253.h>
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 1ca7427c4b6d..bcf2a0698d54 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -561,7 +561,7 @@ endif	# SOUND_OSS
 
 config SOUND_SH_DAC_AUDIO
         tristate "SuperH DAC audio support"
-        depends on CPU_SH3
+        depends on CPU_SH3 && HIGH_RES_TIMERS
 
 config SOUND_SH_DAC_AUDIO_CHANNEL
         int "DAC channel"
diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c
index e4282d93a1aa..21eb6dce46df 100644
--- a/sound/oss/msnd.c
+++ b/sound/oss/msnd.c
@@ -100,7 +100,7 @@ void msnd_fifo_free(msnd_fifo *f)
 int msnd_fifo_alloc(msnd_fifo *f, size_t n)
 {
         msnd_fifo_free(f);
-        f->data = (char *)vmalloc(n);
+        f->data = vmalloc(n);
         f->n = n;
         f->tail = 0;
         f->head = 0;
diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
index 36c3ea62086b..8f7d175767a2 100644
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -17,7 +17,7 @@
 
 #include <linux/init.h>
 #include <linux/spinlock.h>
-#include <asm/timex.h>
+#include <linux/timex.h>
 #include "sound_config.h"
 
 #include "pas2.h"
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
index 78cfb66e4c59..b2ed8757542a 100644
--- a/sound/oss/sh_dac_audio.c
+++ b/sound/oss/sh_dac_audio.c
@@ -18,47 +18,36 @@
 #include <linux/sound.h>
 #include <linux/soundcard.h>
 #include <linux/interrupt.h>
+#include <linux/hrtimer.h>
 #include <asm/io.h>
 #include <asm/uaccess.h>
 #include <asm/irq.h>
 #include <asm/delay.h>
 #include <asm/clock.h>
-#include <asm/cpu/dac.h>
-#include <asm/cpu/timer.h>
+#include <cpu/dac.h>
 #include <asm/machvec.h>
 #include <mach/hp6xx.h>
 #include <asm/hd64461.h>
 
 #define MODNAME "sh_dac_audio"
 
-#define TMU_TOCR_INIT   0x00
-
-#define TMU1_TCR_INIT   0x0020  /* Clock/4, rising edge; interrupt on */
-#define TMU1_TSTR_INIT  0x02    /* Bit to turn on TMU1 */
-
 #define BUFFER_SIZE 48000
 
 static int rate;
 static int empty;
 static char *data_buffer, *buffer_begin, *buffer_end;
 static int in_use, device_major;
+static struct hrtimer hrtimer;
+static ktime_t wakeups_per_second;
 
 static void dac_audio_start_timer(void)
 {
-        u8 tstr;
-
-        tstr = ctrl_inb(TMU_TSTR);
-        tstr |= TMU1_TSTR_INIT;
-        ctrl_outb(tstr, TMU_TSTR);
+        hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL);
 }
 
 static void dac_audio_stop_timer(void)
 {
-        u8 tstr;
-
-        tstr = ctrl_inb(TMU_TSTR);
-        tstr &= ~TMU1_TSTR_INIT;
-        ctrl_outb(tstr, TMU_TSTR);
+        hrtimer_cancel(&hrtimer);
 }
 
 static void dac_audio_reset(void)
@@ -77,38 +66,30 @@ static void dac_audio_sync(void)
 static void dac_audio_start(void)
 {
         if (mach_is_hp6xx()) {
-                u16 v = inw(HD64461_GPADR);
+                u16 v = __raw_readw(HD64461_GPADR);
                 v &= ~HD64461_GPADR_SPEAKER;
-                outw(v, HD64461_GPADR);
+                __raw_writew(v, HD64461_GPADR);
         }
 
         sh_dac_enable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
-        ctrl_outw(TMU1_TCR_INIT, TMU1_TCR);
 }
 static void dac_audio_stop(void)
 {
         dac_audio_stop_timer();
 
         if (mach_is_hp6xx()) {
-                u16 v = inw(HD64461_GPADR);
+                u16 v = __raw_readw(HD64461_GPADR);
                 v |= HD64461_GPADR_SPEAKER;
-                outw(v, HD64461_GPADR);
+                __raw_writew(v, HD64461_GPADR);
         }
 
-        sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
+        sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
         sh_dac_disable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
 }
 
 static void dac_audio_set_rate(void)
 {
-        unsigned long interval;
-        struct clk *clk;
-
-        clk = clk_get(NULL, "module_clk");
-        interval = (clk_get_rate(clk) / 4) / rate;
-        clk_put(clk);
-        ctrl_outl(interval, TMU1_TCOR);
-        ctrl_outl(interval, TMU1_TCNT);
+        wakeups_per_second = ktime_set(0, 1000000000 / rate);
 }
 
 static int dac_audio_ioctl(struct inode *inode, struct file *file,
@@ -265,32 +246,26 @@ const struct file_operations dac_audio_fops = {
       .release =        dac_audio_release,
 };
 
-static irqreturn_t timer1_interrupt(int irq, void *dev)
+static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle)
 {
-        unsigned long timer_status;
-
-        timer_status = ctrl_inw(TMU1_TCR);
-        timer_status &= ~0x100;
-        ctrl_outw(timer_status, TMU1_TCR);
-
         if (!empty) {
                 sh_dac_output(*buffer_begin, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
                 buffer_begin++;
 
                 if (buffer_begin == data_buffer + BUFFER_SIZE)
                         buffer_begin = data_buffer;
-                if (buffer_begin == buffer_end) {
+                if (buffer_begin == buffer_end)
                         empty = 1;
-                        dac_audio_stop_timer();
-                }
         }
-        return IRQ_HANDLED;
+
+        if (!empty)
+                hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL);
+
+        return HRTIMER_NORESTART;
 }
 
 static int __init dac_audio_init(void)
 {
-        int retval;
-
         if ((device_major = register_sound_dsp(&dac_audio_fops, -1)) < 0) {
                 printk(KERN_ERR "Cannot register dsp device");
                 return device_major;
@@ -306,21 +281,25 @@ static int __init dac_audio_init(void)
         rate = 8000;
         dac_audio_set_rate();
 
-        retval =
-            request_irq(TIMER1_IRQ, timer1_interrupt, IRQF_DISABLED, MODNAME, 0);
-        if (retval < 0) {
-                printk(KERN_ERR "sh_dac_audio: IRQ %d request failed\n",
-                       TIMER1_IRQ);
-                return retval;
-        }
+        /* Today: High Resolution Timer driven DAC playback.
+         * The timer callback gets called once per sample. Ouch.
+         *
+         * Future: A much better approach would be to use the
+         * SH7720 CMT+DMAC+DAC hardware combination like this:
+         * - Program sample rate using CMT0 or CMT1
+         * - Program DMAC to use CMT for timing and output to DAC
+         * - Play sound using DMAC, let CPU sleep.
+         * - While at it, rewrite this driver to use ALSA.
+         */
+
+        hrtimer_init(&hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+        hrtimer.function = sh_dac_audio_timer;
 
         return 0;
 }
 
 static void __exit dac_audio_exit(void)
 {
-        free_irq(TIMER1_IRQ, 0);
-
         unregister_sound_dsp(device_major);
         kfree((void *)data_buffer);
 }
diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c
index 6a3891ab69dd..296123ab74f7 100644
--- a/sound/pci/aw2/aw2-saa7146.c
+++ b/sound/pci/aw2/aw2-saa7146.c
@@ -108,7 +108,7 @@ void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip,
 #endif
         /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */
 
-        /* At initialization WS1 and WS2 are disbaled (configured as input */
+        /* At initialization WS1 and WS2 are disabled (configured as input) */
         acon1 |= 0 * WS1_CTRL;
         acon1 |= 0 * WS2_CTRL;
 
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index ccf1b38c88ea..18da2ef04d09 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -988,7 +988,7 @@ static int __devinit snd_lx6464es_create(struct snd_card *card,
         pci_set_master(pci);
 
         /* check if we can restrict PCI DMA transfers to 32 bits */
-        err = pci_set_dma_mask(pci, DMA_32BIT_MASK);
+        err = pci_set_dma_mask(pci, DMA_BIT_MASK(32));
         if (err < 0) {
                 snd_printk(KERN_ERR "architecture does not support "
                            "32bit PCI busmaster DMA\n");
diff --git a/sound/pci/sis7019.h b/sound/pci/sis7019.h
index 013b6739a742..bc8c76819408 100644
--- a/sound/pci/sis7019.h
+++ b/sound/pci/sis7019.h
@@ -203,7 +203,7 @@
 #define SIS_WEISR_B     0xac
 
 
-/* Playback DMA parameters (paramter RAM) */
+/* Playback DMA parameters (parameter RAM) */
 #define SIS_PLAY_DMA_OFFSET     0x0000
 #define SIS_PLAY_DMA_SIZE       0x10
 #define SIS_PLAY_DMA_ADDR(addr, num) \
@@ -228,7 +228,7 @@
 #define         SIS_PLAY_DMA_SSO_MASK           0xffff0000
 #define         SIS_PLAY_DMA_ESO_MASK           0x0000ffff
 
-/* Capture DMA parameters (paramter RAM) */
+/* Capture DMA parameters (parameter RAM) */
 #define SIS_CAPTURE_DMA_OFFSET  0x0800
 #define SIS_CAPTURE_DMA_SIZE    0x10
 #define SIS_CAPTURE_DMA_ADDR(addr, num) \
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index c0efe4491116..6416d3f0c7be 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -367,7 +367,7 @@ static int vx2_load_xilinx_binary(struct vx_core *chip, const struct firmware *x
         unsigned int port;
         const unsigned char *image;
 
-        /* XILINX reset (wait at least 1 milisecond between reset on and off). */
+        /* XILINX reset (wait at least 1 millisecond between reset on and off). */
         vx_outl(chip, CNTRL, VX_CNTRL_REGISTER_VALUE | VX_XILINX_RESET_MASK);
         vx_inl(chip, CNTRL);
         msleep(10);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index dcd163a4ee9a..6375b4ea525d 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -98,13 +98,14 @@ config SND_PXA2XX_SOC_EM_X270
           CompuLab EM-x270, eXeda and CM-X300 machines.
 
 config SND_PXA2XX_SOC_PALM27X
-        bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
-        depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+        bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
+        depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
+                        MACH_PALMT5 || MACH_PALMTE2)
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9712
         help
           Say Y if you want to add support for SoC audio on
-          Palm T|X, T5 or LifeDrive handheld computer.
+          Palm T|X, T5, E2 or LifeDrive handheld computer.
 
 config SND_SOC_ZYLONITE
         tristate "SoC Audio support for Marvell Zylonite"
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 44fcc4e01e08..e6102fda0a7f 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -205,7 +205,7 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
         int ret;
 
         if (!(machine_is_palmtx() || machine_is_palmt5() ||
-                machine_is_palmld()))
+                machine_is_palmld() || machine_is_palmte2()))
                 return -ENODEV;
 
         if (pdev->dev.platform_data)
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 168a088ba761..a587ec40b449 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/delay.h>
+#include <linux/gpio.h>
 #include <linux/clk.h>
 #include <linux/kernel.h>
 #include <linux/io.h>
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 3698f707c44d..3f03d5ddfacd 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -19,6 +19,7 @@
 #include <linux/io.h>
 #include <linux/wait.h>
 #include <linux/delay.h>
+#include <linux/gpio.h>
 #include <linux/clk.h>
 
 #include <sound/core.h>
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index cc066964dad6..556e35f0ab73 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -21,6 +21,8 @@
 #include <linux/clk.h>
 #include <linux/jiffies.h>
 #include <linux/io.h>
+#include <linux/gpio.h>
+
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 169ddad31575..eecfa5eba06b 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -218,24 +218,17 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
          * sync to pclk, half-word transfers to the IIS-FIFO. */
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
                 s3c2410_dma_devconfig(prtd->params->channel,
-                                S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
-                                S3C2410_DISRCC_APB, prtd->params->dma_addr);
-
-                s3c2410_dma_config(prtd->params->channel,
-                                prtd->params->dma_size,
-                                S3C2410_DCON_SYNC_PCLK |
-                                S3C2410_DCON_HANDSHAKE);
+                                      S3C2410_DMASRC_MEM,
+                                      prtd->params->dma_addr);
         } else {
-                s3c2410_dma_config(prtd->params->channel,
-                                prtd->params->dma_size,
-                                S3C2410_DCON_HANDSHAKE |
-                                S3C2410_DCON_SYNC_PCLK);
-
                 s3c2410_dma_devconfig(prtd->params->channel,
-                                        S3C2410_DMASRC_HW, 0x3,
-                                        prtd->params->dma_addr);
+                                      S3C2410_DMASRC_HW,
+                                      prtd->params->dma_addr);
         }
 
+        s3c2410_dma_config(prtd->params->channel,
+                           prtd->params->dma_size);
+
         /* flush the DMA channel */
         s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH);
         prtd->dma_loaded = 0;
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 2b302bbffe73..12522e6913d9 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -27,6 +27,11 @@ MODULE_DESCRIPTION("Core sound module");
 MODULE_AUTHOR("Alan Cox");
 MODULE_LICENSE("GPL");
 
+static char *sound_nodename(struct device *dev)
+{
+        return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev));
+}
+
 static int __init init_soundcore(void)
 {
         int rc;
@@ -41,6 +46,8 @@ static int __init init_soundcore(void)
                 return PTR_ERR(sound_class);
         }
 
+        sound_class->nodename = sound_nodename;
+
         return 0;
 }
 
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index ecb58e7a6245..4bd3a7a0edc1 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -423,7 +423,7 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali
         value_set = convert_bytes_value(cval, value_set);
         buf[0] = value_set & 0xff;
         buf[1] = (value_set >> 8) & 0xff;
-        while (timeout -- > 0)
+        while (timeout-- > 0)
                 if (snd_usb_ctl_msg(cval->mixer->chip->dev,
                                     usb_sndctrlpipe(cval->mixer->chip->dev, 0),
                                     request,