From ecf8c65aba2e406339dc379e7126cc0a7b80ee46 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Fri, 3 Apr 2009 09:41:40 +0200
Subject: [PATCH] --- yaml --- r: 142719 b: refs/heads/master c:
 488fe1665fd90f204fbc825b90060b9e0394e4be h: refs/heads/master i:   142717:
 a8ab9e9c785cece383af1b6530965e010ab60547   142715:
 5bf242f15f7630bbf03ab13be61eafb51d5a1163   142711:
 3d3ff7cf791d86dffd0555a28b28f64fdf3bebd5   142703:
 09069030805d8a653bb40cc9e8fa7412aecae9dd   142687:
 39a6643bf0177911a653a673fa7f24f0516c2439   142655:
 6d910146fca73f4fbbec0a71ee74a863ccca0e98   142591:
 9e5975a3284a2c2c5c5dcfa885e3c8f70b808d2b v: v3

---
 [refs]                                      |   2 +-
 trunk/Documentation/sound/alsa/soc/jack.txt |  71 ---
 trunk/sound/arm/pxa2xx-ac97-lib.c           |  15 +-
 trunk/sound/atmel/abdac.c                   |   4 +-
 trunk/sound/atmel/ac97c.c                   | 128 +----
 trunk/sound/atmel/ac97c.h                   |  14 +-
 trunk/sound/isa/opl3sa2.c                   |   1 -
 trunk/sound/pci/hda/patch_realtek.c         |   5 -
 trunk/sound/pci/hda/patch_sigmatel.c        |   2 -
 trunk/sound/ppc/powermac.c                  |   2 +-
 trunk/sound/soc/codecs/twl4030.c            |  59 +--
 trunk/sound/soc/codecs/twl4030.h            |   1 -
 trunk/sound/soc/codecs/wm9705.c             |  37 --
 trunk/sound/soc/fsl/fsl_dma.c               |  17 -
 trunk/sound/soc/fsl/fsl_ssi.c               |  99 ++--
 trunk/sound/soc/omap/omap-mcbsp.c           |  11 -
 trunk/sound/soc/pxa/Kconfig                 |  10 -
 trunk/sound/soc/pxa/Makefile                |   2 -
 trunk/sound/soc/pxa/magician.c              | 560 --------------------
 trunk/sound/soc/pxa/pxa-ssp.c               |  12 +-
 trunk/sound/soc/soc-core.c                  |  20 +-
 trunk/sound/usb/usbaudio.c                  |   3 +-
 22 files changed, 95 insertions(+), 980 deletions(-)
 delete mode 100644 trunk/Documentation/sound/alsa/soc/jack.txt
 delete mode 100644 trunk/sound/soc/pxa/magician.c

diff --git a/[refs] b/[refs]
index 7eefdec24940..e001e37a00b2 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 5f9c510e9e18cd029e15190d35dd4271f2ef393b
+refs/heads/master: 488fe1665fd90f204fbc825b90060b9e0394e4be
diff --git a/trunk/Documentation/sound/alsa/soc/jack.txt b/trunk/Documentation/sound/alsa/soc/jack.txt
deleted file mode 100644
index fcf82a417293..000000000000
--- a/trunk/Documentation/sound/alsa/soc/jack.txt
+++ /dev/null
@@ -1,71 +0,0 @@
-ASoC jack detection
-===================
-
-ALSA has a standard API for representing physical jacks to user space,
-the kernel side of which can be seen in include/sound/jack.h.  ASoC
-provides a version of this API adding two additional features:
-
- - It allows more than one jack detection method to work together on one
-   user visible jack.  In embedded systems it is common for multiple
-   to be present on a single jack but handled by separate bits of
-   hardware.
-
- - Integration with DAPM, allowing DAPM endpoints to be updated
-   automatically based on the detected jack status (eg, turning off the
-   headphone outputs if no headphones are present).
-
-This is done by splitting the jacks up into three things working
-together: the jack itself represented by a struct snd_soc_jack, sets of
-snd_soc_jack_pins representing DAPM endpoints to update and blocks of
-code providing jack reporting mechanisms.
-
-For example, a system may have a stereo headset jack with two reporting
-mechanisms, one for the headphone and one for the microphone.  Some
-systems won't be able to use their speaker output while a headphone is
-connected and so will want to make sure to update both speaker and
-headphone when the headphone jack status changes.
-
-The jack - struct snd_soc_jack
-==============================
-
-This represents a physical jack on the system and is what is visible to
-user space.  The jack itself is completely passive, it is set up by the
-machine driver and updated by jack detection methods.
-
-Jacks are created by the machine driver calling snd_soc_jack_new().
-
-snd_soc_jack_pin
-================
-
-These represent a DAPM pin to update depending on some of the status
-bits supported by the jack.  Each snd_soc_jack has zero or more of these
-which are updated automatically.  They are created by the machine driver
-and associated with the jack using snd_soc_jack_add_pins().  The status
-of the endpoint may configured to be the opposite of the jack status if
-required (eg, enabling a built in microphone if a microphone is not
-connected via a jack).
-
-Jack detection methods
-======================
-
-Actual jack detection is done by code which is able to monitor some
-input to the system and update a jack by calling snd_soc_jack_report(),
-specifying a subset of bits to update.  The jack detection code should
-be set up by the machine driver, taking configuration for the jack to
-update and the set of things to report when the jack is connected.
-
-Often this is done based on the status of a GPIO - a handler for this is
-provided by the snd_soc_jack_add_gpio() function.  Other methods are
-also available, for example integrated into CODECs.  One example of
-CODEC integrated jack detection can be see in the WM8350 driver.
-
-Each jack may have multiple reporting mechanisms, though it will need at
-least one to be useful.
-
-Machine drivers
-===============
-
-These are all hooked together by the machine driver depending on the
-system hardware.  The machine driver will set up the snd_soc_jack and
-the list of pins to update then set up one or more jack detection
-mechanisms to update that jack based on their current status.
diff --git a/trunk/sound/arm/pxa2xx-ac97-lib.c b/trunk/sound/arm/pxa2xx-ac97-lib.c
index 0afd1a8226fb..7793d2a511ce 100644
--- a/trunk/sound/arm/pxa2xx-ac97-lib.c
+++ b/trunk/sound/arm/pxa2xx-ac97-lib.c
@@ -238,8 +238,6 @@ static inline void pxa_ac97_cold_pxa3xx(void)
 
 bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 {
-	unsigned long gsr;
-
 #ifdef CONFIG_PXA25x
 	if (cpu_is_pxa25x())
 		pxa_ac97_warm_pxa25x();
@@ -256,10 +254,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 	else
 #endif
 		BUG();
-	gsr = GSR | gsr_bits;
-	if (!(gsr & (GSR_PCR | GSR_SCR))) {
+
+	if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
 		printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
-				 __func__, gsr);
+				 __func__, gsr_bits);
 
 		return false;
 	}
@@ -270,8 +268,6 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
 
 bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 {
-	unsigned long gsr;
-
 #ifdef CONFIG_PXA25x
 	if (cpu_is_pxa25x())
 		pxa_ac97_cold_pxa25x();
@@ -289,10 +285,9 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 #endif
 		BUG();
 
-	gsr = GSR | gsr_bits;
-	if (!(gsr & (GSR_PCR | GSR_SCR))) {
+	if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
 		printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
-				 __func__, gsr);
+				 __func__, gsr_bits);
 
 		return false;
 	}
diff --git a/trunk/sound/atmel/abdac.c b/trunk/sound/atmel/abdac.c
index f2f41c854221..28b3c7f7cfe6 100644
--- a/trunk/sound/atmel/abdac.c
+++ b/trunk/sound/atmel/abdac.c
@@ -165,7 +165,7 @@ static struct snd_pcm_hardware atmel_abdac_hw = {
 	.buffer_bytes_max	= 64 * 4096,
 	.period_bytes_min	= 4096,
 	.period_bytes_max	= 4096,
-	.periods_min		= 6,
+	.periods_min		= 4,
 	.periods_max		= 64,
 };
 
@@ -502,7 +502,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, card);
 
 	dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n",
-			dac->regs, dev_name(&dac->dma.chan->dev->device));
+			dac->regs, dac->dma.chan->dev->device.bus_id);
 
 	return retval;
 
diff --git a/trunk/sound/atmel/ac97c.c b/trunk/sound/atmel/ac97c.c
index 0c0f8771656a..dd72e00e5ae1 100644
--- a/trunk/sound/atmel/ac97c.c
+++ b/trunk/sound/atmel/ac97c.c
@@ -1,5 +1,5 @@
 /*
- * Driver for Atmel AC97C
+ * Driver for the Atmel AC97C controller
  *
  * Copyright (C) 2005-2009 Atmel Corporation
  *
@@ -10,7 +10,6 @@
 #include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/bitmap.h>
-#include <linux/device.h>
 #include <linux/dmaengine.h>
 #include <linux/dma-mapping.h>
 #include <linux/init.h>
@@ -66,7 +65,6 @@ struct atmel_ac97c {
 	/* Serialize access to opened variable */
 	spinlock_t			lock;
 	void __iomem			*regs;
-	int				irq;
 	int				opened;
 	int				reset_pin;
 };
@@ -152,10 +150,10 @@ static struct snd_pcm_hardware atmel_ac97c_hw = {
 	.rate_max		= 48000,
 	.channels_min		= 1,
 	.channels_max		= 2,
-	.buffer_bytes_max	= 2 * 2 * 64 * 2048,
+	.buffer_bytes_max	= 64 * 4096,
 	.period_bytes_min	= 4096,
 	.period_bytes_max	= 4096,
-	.periods_min		= 6,
+	.periods_min		= 4,
 	.periods_max		= 64,
 };
 
@@ -299,11 +297,9 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
 {
 	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	unsigned long word = ac97c_readl(chip, OCA);
+	unsigned long word = 0;
 	int retval;
 
-	word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
-
 	/* assign channels to AC97C channel A */
 	switch (runtime->channels) {
 	case 1:
@@ -316,6 +312,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
 	default:
 		/* TODO: support more than two channels */
 		return -EINVAL;
+		break;
 	}
 	ac97c_writel(chip, OCA, word);
 
@@ -327,25 +324,13 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
 		word |= AC97C_CMR_CEM_LITTLE;
 		break;
 	case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+	default:
 		word &= ~(AC97C_CMR_CEM_LITTLE);
 		break;
-	default:
-		word = ac97c_readl(chip, OCA);
-		word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
-		ac97c_writel(chip, OCA, word);
-		return -EINVAL;
 	}
 
-	/* Enable underrun interrupt on channel A */
-	word |= AC97C_CSR_UNRUN;
-
 	ac97c_writel(chip, CAMR, word);
 
-	/* Enable channel A event interrupt */
-	word = ac97c_readl(chip, IMR);
-	word |= AC97C_SR_CAEVT;
-	ac97c_writel(chip, IER, word);
-
 	/* set variable rate if needed */
 	if (runtime->rate != 48000) {
 		word = ac97c_readl(chip, MR);
@@ -374,11 +359,9 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
 {
 	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	unsigned long word = ac97c_readl(chip, ICA);
+	unsigned long word = 0;
 	int retval;
 
-	word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
-
 	/* assign channels to AC97C channel A */
 	switch (runtime->channels) {
 	case 1:
@@ -391,6 +374,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
 	default:
 		/* TODO: support more than two channels */
 		return -EINVAL;
+		break;
 	}
 	ac97c_writel(chip, ICA, word);
 
@@ -402,25 +386,13 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
 		word |= AC97C_CMR_CEM_LITTLE;
 		break;
 	case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+	default:
 		word &= ~(AC97C_CMR_CEM_LITTLE);
 		break;
-	default:
-		word = ac97c_readl(chip, ICA);
-		word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
-		ac97c_writel(chip, ICA, word);
-		return -EINVAL;
 	}
 
-	/* Enable overrun interrupt on channel A */
-	word |= AC97C_CSR_OVRUN;
-
 	ac97c_writel(chip, CAMR, word);
 
-	/* Enable channel A event interrupt */
-	word = ac97c_readl(chip, IMR);
-	word |= AC97C_SR_CAEVT;
-	ac97c_writel(chip, IER, word);
-
 	/* set variable rate if needed */
 	if (runtime->rate != 48000) {
 		word = ac97c_readl(chip, MR);
@@ -571,43 +543,6 @@ static struct snd_pcm_ops atmel_ac97_capture_ops = {
 	.pointer	= atmel_ac97c_capture_pointer,
 };
 
-static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
-{
-	struct atmel_ac97c	*chip  = (struct atmel_ac97c *)dev;
-	irqreturn_t		retval = IRQ_NONE;
-	u32			sr     = ac97c_readl(chip, SR);
-	u32			casr   = ac97c_readl(chip, CASR);
-	u32			cosr   = ac97c_readl(chip, COSR);
-
-	if (sr & AC97C_SR_CAEVT) {
-		dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
-				casr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
-				casr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
-				casr & AC97C_CSR_UNRUN   ? " UNRUN"   : "",
-				casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
-				casr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
-				!casr                    ? " NONE"    : "");
-		retval = IRQ_HANDLED;
-	}
-
-	if (sr & AC97C_SR_COEVT) {
-		dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n",
-				cosr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
-				cosr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
-				cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
-				cosr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
-				!cosr                    ? " NONE"    : "");
-		retval = IRQ_HANDLED;
-	}
-
-	if (retval == IRQ_NONE) {
-		dev_err(&chip->pdev->dev, "spurious interrupt sr 0x%08x "
-				"casr 0x%08x cosr 0x%08x\n", sr, casr, cosr);
-	}
-
-	return retval;
-}
-
 static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip)
 {
 	struct snd_pcm		*pcm;
@@ -730,17 +665,17 @@ static bool filter(struct dma_chan *chan, void *slave)
 
 static void atmel_ac97c_reset(struct atmel_ac97c *chip)
 {
-	ac97c_writel(chip, MR,   0);
-	ac97c_writel(chip, MR,   AC97C_MR_ENA);
-	ac97c_writel(chip, CAMR, 0);
-	ac97c_writel(chip, COMR, 0);
+	ac97c_writel(chip, MR, AC97C_MR_WRST);
 
 	if (gpio_is_valid(chip->reset_pin)) {
 		gpio_set_value(chip->reset_pin, 0);
 		/* AC97 v2.2 specifications says minimum 1 us. */
-		udelay(2);
+		udelay(10);
 		gpio_set_value(chip->reset_pin, 1);
 	}
+
+	udelay(1);
+	ac97c_writel(chip, MR, AC97C_MR_ENA);
 }
 
 static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
@@ -755,7 +690,6 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 		.read	= atmel_ac97c_read,
 	};
 	int				retval;
-	int				irq;
 
 	regs = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!regs) {
@@ -769,12 +703,6 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 		return -ENXIO;
 	}
 
-	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_dbg(&pdev->dev, "could not get irq\n");
-		return -ENXIO;
-	}
-
 	pclk = clk_get(&pdev->dev, "pclk");
 	if (IS_ERR(pclk)) {
 		dev_dbg(&pdev->dev, "no peripheral clock\n");
@@ -791,13 +719,6 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 
 	chip = get_chip(card);
 
-	retval = request_irq(irq, atmel_ac97c_interrupt, 0, "AC97C", chip);
-	if (retval) {
-		dev_dbg(&pdev->dev, "unable to request irq %d\n", irq);
-		goto err_request_irq;
-	}
-	chip->irq = irq;
-
 	spin_lock_init(&chip->lock);
 
 	strcpy(card->driver, "Atmel AC97C");
@@ -826,18 +747,14 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 
 	snd_card_set_dev(card, &pdev->dev);
 
-	atmel_ac97c_reset(chip);
-
-	/* Enable overrun interrupt from codec channel */
-	ac97c_writel(chip, COMR, AC97C_CSR_OVRUN);
-	ac97c_writel(chip, IER, ac97c_readl(chip, IMR) | AC97C_SR_COEVT);
-
 	retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus);
 	if (retval) {
 		dev_dbg(&pdev->dev, "could not register on ac97 bus\n");
 		goto err_ac97_bus;
 	}
 
+	atmel_ac97c_reset(chip);
+
 	retval = atmel_ac97c_mixer_new(chip);
 	if (retval) {
 		dev_dbg(&pdev->dev, "could not register ac97 mixer\n");
@@ -856,7 +773,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 		chip->dma.rx_chan = dma_request_channel(mask, filter, dws);
 
 		dev_info(&chip->pdev->dev, "using %s for DMA RX\n",
-				dev_name(&chip->dma.rx_chan->dev->device));
+					chip->dma.rx_chan->dev->device.bus_id);
 		set_bit(DMA_RX_CHAN_PRESENT, &chip->flags);
 	}
 
@@ -872,7 +789,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 		chip->dma.tx_chan = dma_request_channel(mask, filter, dws);
 
 		dev_info(&chip->pdev->dev, "using %s for DMA TX\n",
-				dev_name(&chip->dma.tx_chan->dev->device));
+					chip->dma.tx_chan->dev->device.bus_id);
 		set_bit(DMA_TX_CHAN_PRESENT, &chip->flags);
 	}
 
@@ -892,7 +809,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 	retval = snd_card_register(card);
 	if (retval) {
 		dev_dbg(&pdev->dev, "could not register sound card\n");
-		goto err_dma;
+		goto err_ac97_bus;
 	}
 
 	platform_set_drvdata(pdev, card);
@@ -919,8 +836,6 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 
 	iounmap(chip->regs);
 err_ioremap:
-	free_irq(irq, chip);
-err_request_irq:
 	snd_card_free(card);
 err_snd_card_new:
 	clk_disable(pclk);
@@ -969,14 +884,9 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev)
 	if (gpio_is_valid(chip->reset_pin))
 		gpio_free(chip->reset_pin);
 
-	ac97c_writel(chip, CAMR, 0);
-	ac97c_writel(chip, COMR, 0);
-	ac97c_writel(chip, MR,   0);
-
 	clk_disable(chip->pclk);
 	clk_put(chip->pclk);
 	iounmap(chip->regs);
-	free_irq(chip->irq, chip);
 
 	if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags))
 		dma_release_channel(chip->dma.rx_chan);
diff --git a/trunk/sound/atmel/ac97c.h b/trunk/sound/atmel/ac97c.h
index ecbba5021c80..c17bd5825980 100644
--- a/trunk/sound/atmel/ac97c.h
+++ b/trunk/sound/atmel/ac97c.h
@@ -1,5 +1,5 @@
 /*
- * Register definitions for Atmel AC97C
+ * Register definitions for the Atmel AC97C controller
  *
  * Copyright (C) 2005-2009 Atmel Corporation
  *
@@ -17,6 +17,10 @@
 #define AC97C_CATHR		0x24
 #define AC97C_CASR		0x28
 #define AC97C_CAMR		0x2c
+#define AC97C_CBRHR		0x30
+#define AC97C_CBTHR		0x34
+#define AC97C_CBSR		0x38
+#define AC97C_CBMR		0x3c
 #define AC97C_CORHR		0x40
 #define AC97C_COTHR		0x44
 #define AC97C_COSR		0x48
@@ -42,10 +46,8 @@
 #define AC97C_MR_VRA		(1 << 2)
 
 #define AC97C_CSR_TXRDY		(1 << 0)
-#define AC97C_CSR_TXEMPTY	(1 << 1)
 #define AC97C_CSR_UNRUN		(1 << 2)
 #define AC97C_CSR_RXRDY		(1 << 4)
-#define AC97C_CSR_OVRUN		(1 << 5)
 #define AC97C_CSR_ENDTX		(1 << 10)
 #define AC97C_CSR_ENDRX		(1 << 14)
 
@@ -59,15 +61,11 @@
 #define AC97C_CMR_DMAEN		(1 << 22)
 
 #define AC97C_SR_CAEVT		(1 << 3)
-#define AC97C_SR_COEVT		(1 << 2)
-#define AC97C_SR_WKUP		(1 << 1)
-#define AC97C_SR_SOF		(1 << 0)
 
-#define AC97C_CH_MASK(slot)						\
-	(0x7 << (3 * (AC97_SLOT_##slot - 3)))
 #define AC97C_CH_ASSIGN(slot, channel)					\
 	(AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3)))
 #define AC97C_CHANNEL_NONE	0x0
 #define AC97C_CHANNEL_A		0x1
+#define AC97C_CHANNEL_B		0x2
 
 #endif /* __SOUND_ATMEL_AC97C_H */
diff --git a/trunk/sound/isa/opl3sa2.c b/trunk/sound/isa/opl3sa2.c
index 0481a55334b9..ef95279da7a3 100644
--- a/trunk/sound/isa/opl3sa2.c
+++ b/trunk/sound/isa/opl3sa2.c
@@ -481,7 +481,6 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1,
 OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1),
 OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1,
 		   db_scale_5bit_12db_max),
-OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0),
 };
 
 static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = {
diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c
index f35e58a2d921..82097790f6f3 100644
--- a/trunk/sound/pci/hda/patch_realtek.c
+++ b/trunk/sound/pci/hda/patch_realtek.c
@@ -8764,10 +8764,6 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	{}
 };
 
-static hda_nid_t alc883_slave_dig_outs[] = {
-	ALC1200_DIGOUT_NID, 0,
-};
-
 static hda_nid_t alc1200_slave_dig_outs[] = {
 	ALC883_DIGOUT_NID, 0,
 };
@@ -8813,7 +8809,6 @@ static struct alc_config_preset alc883_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
 		.dig_in_nid = ALC883_DIGIN_NID,
-		.slave_dig_outs = alc883_slave_dig_outs,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
 		.channel_mode = alc883_3ST_6ch_intel_modes,
 		.need_dac_fix = 1,
diff --git a/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index b34d78b88a85..b5e108aa8f63 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/sound/pci/hda/patch_sigmatel.c
@@ -4895,7 +4895,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	switch (codec->vendor_id) {
 	case 0x111d7604:
 	case 0x111d7605:
-	case 0x111d76d5:
 		if (spec->board_config == STAC_92HD83XXX_PWR_REF)
 			break;
 		spec->num_pwrs = 0;
@@ -5708,7 +5707,6 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
diff --git a/trunk/sound/ppc/powermac.c b/trunk/sound/ppc/powermac.c
index a2b69b8cff43..5a929069dce9 100644
--- a/trunk/sound/ppc/powermac.c
+++ b/trunk/sound/ppc/powermac.c
@@ -51,7 +51,7 @@ static struct platform_device *device;
 /*
  */
 
-static int __devinit snd_pmac_probe(struct platform_device *devptr)
+static int __init snd_pmac_probe(struct platform_device *devptr)
 {
 	struct snd_card *card;
 	struct snd_pmac *chip;
diff --git a/trunk/sound/soc/codecs/twl4030.c b/trunk/sound/soc/codecs/twl4030.c
index bfda7a88e825..97738e2ece04 100644
--- a/trunk/sound/soc/codecs/twl4030.c
+++ b/trunk/sound/soc/codecs/twl4030.c
@@ -122,9 +122,6 @@ struct twl4030_priv {
 	unsigned int bypass_state;
 	unsigned int codec_powered;
 	unsigned int codec_muted;
-
-	struct snd_pcm_substream *master_substream;
-	struct snd_pcm_substream *slave_substream;
 };
 
 /*
@@ -1220,50 +1217,6 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int twl4030_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
-
-	/* If we already have a playback or capture going then constrain
-	 * this substream to match it.
-	 */
-	if (twl4030->master_substream) {
-		struct snd_pcm_runtime *master_runtime;
-		master_runtime = twl4030->master_substream->runtime;
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_RATE,
-					     master_runtime->rate,
-					     master_runtime->rate);
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     master_runtime->sample_bits,
-					     master_runtime->sample_bits);
-
-		twl4030->slave_substream = substream;
-	} else
-		twl4030->master_substream = substream;
-
-	return 0;
-}
-
-static void twl4030_shutdown(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
-
-	if (twl4030->master_substream == substream)
-		twl4030->master_substream = twl4030->slave_substream;
-
-	twl4030->slave_substream = NULL;
-}
-
 static int twl4030_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
@@ -1271,13 +1224,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
 	u8 mode, old_mode, format, old_format;
 
-	if (substream == twl4030->slave_substream)
-		/* Ignoring hw_params for slave substream */
-		return 0;
-
 	/* bit rate */
 	old_mode = twl4030_read_reg_cache(codec,
 			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1311,9 +1259,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	case 48000:
 		mode |= TWL4030_APLL_RATE_48000;
 		break;
-	case 96000:
-		mode |= TWL4030_APLL_RATE_96000;
-		break;
 	default:
 		printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
 			params_rate(params));
@@ -1439,8 +1384,6 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
 static struct snd_soc_dai_ops twl4030_dai_ops = {
-	.startup	= twl4030_startup,
-	.shutdown	= twl4030_shutdown,
 	.hw_params	= twl4030_hw_params,
 	.set_sysclk	= twl4030_set_dai_sysclk,
 	.set_fmt	= twl4030_set_dai_fmt,
@@ -1452,7 +1395,7 @@ struct snd_soc_dai twl4030_dai = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
+		.rates = TWL4030_RATES,
 		.formats = TWL4030_FORMATS,},
 	.capture = {
 		.stream_name = "Capture",
diff --git a/trunk/sound/soc/codecs/twl4030.h b/trunk/sound/soc/codecs/twl4030.h
index cb63765db1df..33dbb144dad1 100644
--- a/trunk/sound/soc/codecs/twl4030.h
+++ b/trunk/sound/soc/codecs/twl4030.h
@@ -109,7 +109,6 @@
 #define TWL4030_APLL_RATE_32000		0x80
 #define TWL4030_APLL_RATE_44100		0x90
 #define TWL4030_APLL_RATE_48000		0xA0
-#define TWL4030_APLL_RATE_96000		0xE0
 #define TWL4030_SEL_16K			0x04
 #define TWL4030_CODECPDZ		0x02
 #define TWL4030_OPT_MODE		0x01
diff --git a/trunk/sound/soc/codecs/wm9705.c b/trunk/sound/soc/codecs/wm9705.c
index 6e23a81dba78..3265817c5c26 100644
--- a/trunk/sound/soc/codecs/wm9705.c
+++ b/trunk/sound/soc/codecs/wm9705.c
@@ -317,41 +317,6 @@ static int wm9705_reset(struct snd_soc_codec *codec)
 	return -EIO;
 }
 
-#ifdef CONFIG_PM
-static int wm9705_soc_suspend(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
-
-	return 0;
-}
-
-static int wm9705_soc_resume(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int i, ret;
-	u16 *cache = codec->reg_cache;
-
-	ret = wm9705_reset(codec);
-	if (ret < 0) {
-		printk(KERN_ERR "could not reset AC97 codec\n");
-		return ret;
-	}
-
-	for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
-		soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
-	}
-
-	return 0;
-}
-#else
-#define wm9705_soc_suspend NULL
-#define wm9705_soc_resume NULL
-#endif
-
 static int wm9705_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -442,8 +407,6 @@ static int wm9705_soc_remove(struct platform_device *pdev)
 struct snd_soc_codec_device soc_codec_dev_wm9705 = {
 	.probe = 	wm9705_soc_probe,
 	.remove = 	wm9705_soc_remove,
-	.suspend =	wm9705_soc_suspend,
-	.resume =	wm9705_soc_resume,
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
 
diff --git a/trunk/sound/soc/fsl/fsl_dma.c b/trunk/sound/soc/fsl/fsl_dma.c
index 2c4892c853cf..b3eb8570cd7b 100644
--- a/trunk/sound/soc/fsl/fsl_dma.c
+++ b/trunk/sound/soc/fsl/fsl_dma.c
@@ -697,23 +697,6 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
 	else
 		position = in_be32(&dma_channel->dar);
 
-	/*
-	 * When capture is started, the SSI immediately starts to fill its FIFO.
-	 * This means that the DMA controller is not started until the FIFO is
-	 * full.  However, ALSA calls this function before that happens, when
-	 * MR.DAR is still zero.  In this case, just return zero to indicate
-	 * that nothing has been received yet.
-	 */
-	if (!position)
-		return 0;
-
-	if ((position < dma_private->dma_buf_phys) ||
-	    (position > dma_private->dma_buf_end)) {
-		dev_err(substream->pcm->card->dev,
-			"dma pointer is out of range, halting stream\n");
-		return SNDRV_PCM_POS_XRUN;
-	}
-
 	frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
 
 	/*
diff --git a/trunk/sound/soc/fsl/fsl_ssi.c b/trunk/sound/soc/fsl/fsl_ssi.c
index 3711d8454d96..169bca295b78 100644
--- a/trunk/sound/soc/fsl/fsl_ssi.c
+++ b/trunk/sound/soc/fsl/fsl_ssi.c
@@ -60,13 +60,6 @@
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
 #endif
 
-/* SIER bitflag of interrupts to enable */
-#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
-		    CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
-		    CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
-		    CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
-		    CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
-
 /**
  * fsl_ssi_private: per-SSI private data
  *
@@ -147,7 +140,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	   were interrupted for.  We mask it with the Interrupt Enable register
 	   so that we only check for events that we're interested in.
 	 */
-	sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
+	sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier);
 
 	if (sisr & CCSR_SSI_SISR_RFRC) {
 		ssi_private->stats.rfrc++;
@@ -331,7 +324,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 */
 
 		/* 4. Enable the interrupts and DMA requests */
-		out_be32(&ssi->sier, SIER_FLAGS);
+		out_be32(&ssi->sier,
+			 CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE |
+			 CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN |
+			 CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN |
+			 CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE |
+			 CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN);
 
 		/*
 		 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We
@@ -468,12 +466,28 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_START:
 		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			setbits32(&ssi->scr,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
-		else
+		} else {
+			long timeout = jiffies + 10;
+
 			setbits32(&ssi->scr,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
+
+			/* Wait until the SSI has filled its FIFO. Without this
+			 * delay, ALSA complains about overruns.  When the FIFO
+			 * is full, the DMA controller initiates its first
+			 * transfer.  Until then, however, the DMA's DAR
+			 * register is zero, which translates to an
+			 * out-of-bounds pointer.  This makes ALSA think an
+			 * overrun has occurred.
+			 */
+			while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
+			       (jiffies < timeout));
+			if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
+				return -EIO;
+		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -592,52 +606,39 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 	.ops = &fsl_ssi_dai_ops,
 };
 
-/* Show the statistics of a flag only if its interrupt is enabled.  The
- * compiler will optimze this code to a no-op if the interrupt is not
- * enabled.
- */
-#define SIER_SHOW(flag, name) \
-	do { \
-		if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
-			length += sprintf(buf + length, #name "=%u\n", \
-				ssi_private->stats.name); \
-	} while (0)
-
-
 /**
  * fsl_sysfs_ssi_show: display SSI statistics
  *
- * Display the statistics for the current SSI device.  To avoid confusion,
- * we only show those counts that are enabled.
+ * Display the statistics for the current SSI device.
  */
 static ssize_t fsl_sysfs_ssi_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct fsl_ssi_private *ssi_private =
-		container_of(attr, struct fsl_ssi_private, dev_attr);
-	ssize_t length = 0;
-
-	SIER_SHOW(RFRC_EN, rfrc);
-	SIER_SHOW(TFRC_EN, tfrc);
-	SIER_SHOW(CMDAU_EN, cmdau);
-	SIER_SHOW(CMDDU_EN, cmddu);
-	SIER_SHOW(RXT_EN, rxt);
-	SIER_SHOW(RDR1_EN, rdr1);
-	SIER_SHOW(RDR0_EN, rdr0);
-	SIER_SHOW(TDE1_EN, tde1);
-	SIER_SHOW(TDE0_EN, tde0);
-	SIER_SHOW(ROE1_EN, roe1);
-	SIER_SHOW(ROE0_EN, roe0);
-	SIER_SHOW(TUE1_EN, tue1);
-	SIER_SHOW(TUE0_EN, tue0);
-	SIER_SHOW(TFS_EN, tfs);
-	SIER_SHOW(RFS_EN, rfs);
-	SIER_SHOW(TLS_EN, tls);
-	SIER_SHOW(RLS_EN, rls);
-	SIER_SHOW(RFF1_EN, rff1);
-	SIER_SHOW(RFF0_EN, rff0);
-	SIER_SHOW(TFE1_EN, tfe1);
-	SIER_SHOW(TFE0_EN, tfe0);
+	container_of(attr, struct fsl_ssi_private, dev_attr);
+	ssize_t length;
+
+	length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc);
+	length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc);
+	length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau);
+	length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu);
+	length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt);
+	length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1);
+	length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0);
+	length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1);
+	length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0);
+	length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1);
+	length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0);
+	length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1);
+	length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0);
+	length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs);
+	length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs);
+	length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls);
+	length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls);
+	length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1);
+	length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0);
+	length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1);
+	length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0);
 
 	return length;
 }
diff --git a/trunk/sound/soc/omap/omap-mcbsp.c b/trunk/sound/soc/omap/omap-mcbsp.c
index 9c09b94f0cf8..d6882be33452 100644
--- a/trunk/sound/soc/omap/omap-mcbsp.c
+++ b/trunk/sound/soc/omap/omap-mcbsp.c
@@ -146,17 +146,6 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	int err = 0;
 
-	if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
-		/*
-		 * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
-		 * Set constraint for minimum buffer size to the same than FIFO
-		 * size in order to avoid underruns in playback startup because
-		 * HW is keeping the DMA request active until FIFO is filled.
-		 */
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
-	}
-
 	if (!cpu_dai->active)
 		err = omap_mcbsp_request(mcbsp_data->bus_id);
 
diff --git a/trunk/sound/soc/pxa/Kconfig b/trunk/sound/soc/pxa/Kconfig
index ad8a10fe6298..5998ab366e83 100644
--- a/trunk/sound/soc/pxa/Kconfig
+++ b/trunk/sound/soc/pxa/Kconfig
@@ -116,16 +116,6 @@ config SND_SOC_ZYLONITE
 	  Say Y if you want to add support for SoC audio on the
 	  Marvell Zylonite reference platform.
 
-config SND_PXA2XX_SOC_MAGICIAN
-	tristate "SoC Audio support for HTC Magician"
-	depends on SND_PXA2XX_SOC && MACH_MAGICIAN
-	select SND_PXA2XX_SOC_I2S
-	select SND_PXA_SOC_SSP
-	select SND_SOC_UDA1380
-	help
-	  Say Y if you want to add support for SoC audio on the
-	  HTC Magician.
-
 config SND_PXA2XX_SOC_MIOA701
         tristate "SoC Audio support for MIO A701"
         depends on SND_PXA2XX_SOC && MACH_MIOA701
diff --git a/trunk/sound/soc/pxa/Makefile b/trunk/sound/soc/pxa/Makefile
index 4b90c3ccae45..8ed881c5e5cc 100644
--- a/trunk/sound/soc/pxa/Makefile
+++ b/trunk/sound/soc/pxa/Makefile
@@ -20,7 +20,6 @@ snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
-snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
@@ -32,6 +31,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
-obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/trunk/sound/soc/pxa/magician.c b/trunk/sound/soc/pxa/magician.c
deleted file mode 100644
index f7c4544f7859..000000000000
--- a/trunk/sound/soc/pxa/magician.c
+++ /dev/null
@@ -1,560 +0,0 @@
-/*
- * SoC audio for HTC Magician
- *
- * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
- *
- * based on spitz.c,
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
- *          Richard Purdie <richard@openedhand.com>
- *
- *  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/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/delay.h>
-#include <linux/gpio.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
-#include <mach/magician.h>
-#include <asm/mach-types.h>
-#include "../codecs/uda1380.h"
-#include "pxa2xx-pcm.h"
-#include "pxa2xx-i2s.h"
-#include "pxa-ssp.h"
-
-#define MAGICIAN_MIC       0
-#define MAGICIAN_MIC_EXT   1
-
-static int magician_hp_switch;
-static int magician_spk_switch = 1;
-static int magician_in_sel = MAGICIAN_MIC;
-
-static void magician_ext_control(struct snd_soc_codec *codec)
-{
-	if (magician_spk_switch)
-		snd_soc_dapm_enable_pin(codec, "Speaker");
-	else
-		snd_soc_dapm_disable_pin(codec, "Speaker");
-	if (magician_hp_switch)
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	else
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-
-	switch (magician_in_sel) {
-	case MAGICIAN_MIC:
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_enable_pin(codec, "Call Mic");
-		break;
-	case MAGICIAN_MIC_EXT:
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
-		snd_soc_dapm_enable_pin(codec, "Headset Mic");
-		break;
-	}
-
-	snd_soc_dapm_sync(codec);
-}
-
-static int magician_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->card->codec;
-
-	/* check the jack status at stream startup */
-	magician_ext_control(codec);
-
-	return 0;
-}
-
-/*
- * Magician uses SSP port for playback.
- */
-static int magician_playback_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;
-	unsigned int acps, acds, width, rate;
-	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
-	int ret = 0;
-
-	rate = params_rate(params);
-	width = snd_pcm_format_physical_width(params_format(params));
-
-	/*
-	 * rate = SSPSCLK / (2 * width(16 or 32))
-	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
-	 */
-	switch (params_rate(params)) {
-	case 8000:
-		/* off by a factor of 2: bug in the PXA27x audio clock? */
-		acps = 32842000;
-		switch (width) {
-		case 16:
-			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_16;
-			break;
-		case 32:
-			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_8;
-		}
-		break;
-	case 11025:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_4;
-			break;
-		case 32:
-			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-		}
-		break;
-	case 22050:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		case 32:
-			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 44100:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		case 32:
-			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 48000:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		case 32:
-			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 96000:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-			break;
-		case 32:
-			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			div4 = PXA_SSP_CLK_SCDB_1;
-			break;
-		}
-		break;
-	}
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
-			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-			SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
-	if (ret < 0)
-		return ret;
-
-	/* set audio clock as clock source */
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
-			SND_SOC_CLOCK_OUT);
-	if (ret < 0)
-		return ret;
-
-	/* set the SSP audio system clock ACDS divider */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_ACDS, acds);
-	if (ret < 0)
-		return ret;
-
-	/* set the SSP audio system clock SCDB divider4 */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_SCDB, div4);
-	if (ret < 0)
-		return ret;
-
-	/* set SSP audio pll clock */
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-/*
- * Magician uses I2S for capture.
- */
-static int magician_capture_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 = 0;
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai,
-			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
-			SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
-			SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set the I2S system clock as output */
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
-			SND_SOC_CLOCK_OUT);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static struct snd_soc_ops magician_capture_ops = {
-	.startup = magician_startup,
-	.hw_params = magician_capture_hw_params,
-};
-
-static struct snd_soc_ops magician_playback_ops = {
-	.startup = magician_startup,
-	.hw_params = magician_playback_hw_params,
-};
-
-static int magician_get_hp(struct snd_kcontrol *kcontrol,
-			     struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = magician_hp_switch;
-	return 0;
-}
-
-static int magician_set_hp(struct snd_kcontrol *kcontrol,
-			     struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-
-	if (magician_hp_switch == ucontrol->value.integer.value[0])
-		return 0;
-
-	magician_hp_switch = ucontrol->value.integer.value[0];
-	magician_ext_control(codec);
-	return 1;
-}
-
-static int magician_get_spk(struct snd_kcontrol *kcontrol,
-			    struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = magician_spk_switch;
-	return 0;
-}
-
-static int magician_set_spk(struct snd_kcontrol *kcontrol,
-			    struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-
-	if (magician_spk_switch == ucontrol->value.integer.value[0])
-		return 0;
-
-	magician_spk_switch = ucontrol->value.integer.value[0];
-	magician_ext_control(codec);
-	return 1;
-}
-
-static int magician_get_input(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = magician_in_sel;
-	return 0;
-}
-
-static int magician_set_input(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol)
-{
-	if (magician_in_sel == ucontrol->value.integer.value[0])
-		return 0;
-
-	magician_in_sel = ucontrol->value.integer.value[0];
-
-	switch (magician_in_sel) {
-	case MAGICIAN_MIC:
-		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
-		break;
-	case MAGICIAN_MIC_EXT:
-		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
-	}
-
-	return 1;
-}
-
-static int magician_spk_power(struct snd_soc_dapm_widget *w,
-				struct snd_kcontrol *k, int event)
-{
-	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
-	return 0;
-}
-
-static int magician_hp_power(struct snd_soc_dapm_widget *w,
-				struct snd_kcontrol *k, int event)
-{
-	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
-	return 0;
-}
-
-static int magician_mic_bias(struct snd_soc_dapm_widget *w,
-				struct snd_kcontrol *k, int event)
-{
-	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
-	return 0;
-}
-
-/* magician machine dapm widgets */
-static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
-	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
-	SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
-	SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
-};
-
-/* magician machine audio_map */
-static const struct snd_soc_dapm_route audio_map[] = {
-
-	/* Headphone connected to VOUTL, VOUTR */
-	{"Headphone Jack", NULL, "VOUTL"},
-	{"Headphone Jack", NULL, "VOUTR"},
-
-	/* Speaker connected to VOUTL, VOUTR */
-	{"Speaker", NULL, "VOUTL"},
-	{"Speaker", NULL, "VOUTR"},
-
-	/* Mics are connected to VINM */
-	{"VINM", NULL, "Headset Mic"},
-	{"VINM", NULL, "Call Mic"},
-};
-
-static const char *input_select[] = {"Call Mic", "Headset Mic"};
-static const struct soc_enum magician_in_sel_enum =
-	SOC_ENUM_SINGLE_EXT(2, input_select);
-
-static const struct snd_kcontrol_new uda1380_magician_controls[] = {
-	SOC_SINGLE_BOOL_EXT("Headphone Switch",
-			(unsigned long)&magician_hp_switch,
-			magician_get_hp, magician_set_hp),
-	SOC_SINGLE_BOOL_EXT("Speaker Switch",
-			(unsigned long)&magician_spk_switch,
-			magician_get_spk, magician_set_spk),
-	SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
-			magician_get_input, magician_set_input),
-};
-
-/*
- * Logic for a uda1380 as connected on a HTC Magician
- */
-static int magician_uda1380_init(struct snd_soc_codec *codec)
-{
-	int err;
-
-	/* NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "VOUTLHP");
-	snd_soc_dapm_nc_pin(codec, "VOUTRHP");
-
-	/* FIXME: is anything connected here? */
-	snd_soc_dapm_nc_pin(codec, "VINL");
-	snd_soc_dapm_nc_pin(codec, "VINR");
-
-	/* Add magician specific controls */
-	err = snd_soc_add_controls(codec, uda1380_magician_controls,
-				ARRAY_SIZE(uda1380_magician_controls));
-	if (err < 0)
-		return err;
-
-	/* Add magician specific widgets */
-	snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
-				  ARRAY_SIZE(uda1380_dapm_widgets));
-
-	/* Set up magician specific audio path interconnects */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
-
-	snd_soc_dapm_sync(codec);
-	return 0;
-}
-
-/* magician digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link magician_dai[] = {
-{
-	.name = "uda1380",
-	.stream_name = "UDA1380 Playback",
-	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
-	.codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
-	.init = magician_uda1380_init,
-	.ops = &magician_playback_ops,
-},
-{
-	.name = "uda1380",
-	.stream_name = "UDA1380 Capture",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
-	.ops = &magician_capture_ops,
-}
-};
-
-/* magician audio machine driver */
-static struct snd_soc_card snd_soc_card_magician = {
-	.name = "Magician",
-	.dai_link = magician_dai,
-	.num_links = ARRAY_SIZE(magician_dai),
-	.platform = &pxa2xx_soc_platform,
-};
-
-/* magician audio private data */
-static struct uda1380_setup_data magician_uda1380_setup = {
-	.i2c_address = 0x18,
-	.dac_clk = UDA1380_DAC_CLK_WSPLL,
-};
-
-/* magician audio subsystem */
-static struct snd_soc_device magician_snd_devdata = {
-	.card = &snd_soc_card_magician,
-	.codec_dev = &soc_codec_dev_uda1380,
-	.codec_data = &magician_uda1380_setup,
-};
-
-static struct platform_device *magician_snd_device;
-
-static int __init magician_init(void)
-{
-	int ret;
-
-	if (!machine_is_magician())
-		return -ENODEV;
-
-	ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
-	if (ret)
-		goto err_request_power;
-	ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
-	if (ret)
-		goto err_request_reset;
-	ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
-	if (ret)
-		goto err_request_spk;
-	ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
-	if (ret)
-		goto err_request_ep;
-	ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
-	if (ret)
-		goto err_request_mic;
-	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
-	if (ret)
-		goto err_request_in_sel0;
-	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
-	if (ret)
-		goto err_request_in_sel1;
-
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
-	gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
-
-	/* we may need to have the clock running here - pH5 */
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
-	udelay(5);
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
-
-	magician_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!magician_snd_device) {
-		ret = -ENOMEM;
-		goto err_pdev;
-	}
-
-	platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
-	magician_snd_devdata.dev = &magician_snd_device->dev;
-	ret = platform_device_add(magician_snd_device);
-	if (ret) {
-		platform_device_put(magician_snd_device);
-		goto err_pdev;
-	}
-
-	return 0;
-
-err_pdev:
-	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
-err_request_in_sel1:
-	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
-err_request_in_sel0:
-	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
-err_request_mic:
-	gpio_free(EGPIO_MAGICIAN_EP_POWER);
-err_request_ep:
-	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
-err_request_spk:
-	gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
-err_request_reset:
-	gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
-err_request_power:
-	return ret;
-}
-
-static void __exit magician_exit(void)
-{
-	platform_device_unregister(magician_snd_device);
-
-	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
-	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
-	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
-
-	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
-	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
-	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
-	gpio_free(EGPIO_MAGICIAN_EP_POWER);
-	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
-	gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
-	gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
-}
-
-module_init(magician_init);
-module_exit(magician_exit);
-
-MODULE_AUTHOR("Philipp Zabel");
-MODULE_DESCRIPTION("ALSA SoC Magician");
-MODULE_LICENSE("GPL");
diff --git a/trunk/sound/soc/pxa/pxa-ssp.c b/trunk/sound/soc/pxa/pxa-ssp.c
index 308a657928d2..7acd3febf8b0 100644
--- a/trunk/sound/soc/pxa/pxa-ssp.c
+++ b/trunk/sound/soc/pxa/pxa-ssp.c
@@ -627,18 +627,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	u32 sscr0;
 	u32 sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
-	int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
 
 	/* select correct DMA params */
 	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
 		dma = 1; /* capture DMA offset is 1,3 */
-	/* Network mode with one active slot (ttsa == 1) can be used
-	 * to force 16-bit frame width on the wire (for S16_LE), even
-	 * with two channels. Use 16-bit DMA transfers for this case.
-	 */
-	if (((chn == 2) && (ttsa != 1)) || (width == 32))
-		dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
-
+	if (chn == 2)
+		dma += 2; /* stereo DMA offset is 2, mono is 0 */
 	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
 
 	dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
@@ -718,7 +712,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	/* When we use a network mode, we always require TDM slots
 	 * - complain loudly and fail if they've not been set up yet.
 	 */
-	if ((sscr0 & SSCR0_MOD) && !ttsa) {
+	if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
 		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
 		return -EINVAL;
 	}
diff --git a/trunk/sound/soc/soc-core.c b/trunk/sound/soc/soc-core.c
index 99712f652d0d..6e710f705a74 100644
--- a/trunk/sound/soc/soc-core.c
+++ b/trunk/sound/soc/soc-core.c
@@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 	int err;
 
 	codec->ac97->dev.bus = &ac97_bus_type;
-	codec->ac97->dev.parent = codec->card->dev;
+	codec->ac97->dev.parent = NULL;
 	codec->ac97->dev.release = soc_ac97_device_release;
 
 	dev_set_name(&codec->ac97->dev, "%d-%d:%s",
@@ -767,21 +767,11 @@ static int soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
 
-	/* AC97 devices might have other drivers hanging off them so
-	 * need to resume immediately.  Other drivers don't have that
-	 * problem and may take a substantial amount of time to resume
-	 * due to I/O costs and anti-pop so handle them out of line.
-	 */
-	if (cpu_dai->ac97_control) {
-		dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
-		soc_resume_deferred(&card->deferred_resume_work);
-	} else {
-		dev_dbg(socdev->dev, "Scheduling resume work\n");
-		if (!schedule_work(&card->deferred_resume_work))
-			dev_err(socdev->dev, "resume work item may be lost\n");
-	}
+	dev_dbg(socdev->dev, "scheduling resume work\n");
+
+	if (!schedule_work(&card->deferred_resume_work))
+		dev_err(socdev->dev, "resume work item may be lost\n");
 
 	return 0;
 }
diff --git a/trunk/sound/usb/usbaudio.c b/trunk/sound/usb/usbaudio.c
index c2db0f959681..175c7d1da5cf 100644
--- a/trunk/sound/usb/usbaudio.c
+++ b/trunk/sound/usb/usbaudio.c
@@ -2147,7 +2147,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
 		fp = list_entry(p, struct audioformat, list);
 		snd_iprintf(buffer, "  Interface %d\n", fp->iface);
 		snd_iprintf(buffer, "    Altset %d\n", fp->altsetting);
-		snd_iprintf(buffer, "    Format: %#x\n", fp->format);
+		snd_iprintf(buffer, "    Format: %#x (%d bits)\n",
+			    fp->format, snd_pcm_format_width(fp->format));
 		snd_iprintf(buffer, "    Channels: %d\n", fp->channels);
 		snd_iprintf(buffer, "    Endpoint: %d %s (%s)\n",
 			    fp->endpoint & USB_ENDPOINT_NUMBER_MASK,