Quellcode durchsuchen

Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Most of changes are fairly small and driver-specific.

  A remaining regression fix for USB-audio sync pipe check, a fix for
  HD-audio power-up sequence, fixes for ASoC pxa-ssp compile issues, and
  bunch of ASoC codec and trivial fix patches."

* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: 6fire: use NULL instead of 0 for pointer assignment
  ALSA: hda - Handle open while transitioning to D3.
  ALSA: snd-usb: make snd_usb_substream_capture_trigger static
  ALSA: snd-usb: fix sync pipe check
  ASoC: tegra+wm8903: turn of mic detect when card is removed
  ASoC: wm8996: Mark the CODEC as cache only when powering off on boot
  ASoC: wm8996: Move reset before the initial regulator disable
  ASoC: wm8996: Remove spurious regulator_bulk_free()
  ASoC: wm8904: Fix cache only management
  ASoC: wm8904: Fix GPIO and MICBIAS initialisation for regmap conversion
  ASoC: fix pxa-ssp compiling issue under mach-mmp
  ARM: MMP: add pxa910-ssp into ssp_id_table
Linus Torvalds vor 13 Jahren
Ursprung
Commit
f40759e704

+ 1 - 0
arch/arm/plat-pxa/ssp.c

@@ -193,6 +193,7 @@ static const struct platform_device_id ssp_id_table[] = {
 	{ "pxa25x-nssp",	PXA25x_NSSP },
 	{ "pxa27x-ssp",		PXA27x_SSP },
 	{ "pxa168-ssp",		PXA168_SSP },
+	{ "pxa910-ssp",		PXA910_SSP },
 	{ },
 };
 

+ 2 - 0
include/linux/pxa2xx_ssp.h

@@ -160,7 +160,9 @@ enum pxa_ssp_type {
 	PXA25x_SSP,  /* pxa 210, 250, 255, 26x */
 	PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */
 	PXA27x_SSP,
+	PXA3xx_SSP,
 	PXA168_SSP,
+	PXA910_SSP,
 	CE4100_SSP,
 };
 

+ 1 - 1
include/linux/spi/pxa2xx_spi.h

@@ -43,7 +43,7 @@ struct pxa2xx_spi_chip {
 	void (*cs_control)(u32 command);
 };
 
-#ifdef CONFIG_ARCH_PXA
+#if defined(CONFIG_ARCH_PXA) || defined(CONFIG_ARCH_MMP)
 
 #include <linux/clk.h>
 #include <mach/dma.h>

+ 37 - 9
sound/pci/hda/hda_codec.c

@@ -4393,20 +4393,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec)
 	codec->power_jiffies += delta;
 }
 
-/**
- * snd_hda_power_up - Power-up the codec
- * @codec: HD-audio codec
- *
- * Increment the power-up counter and power up the hardware really when
- * not turned on yet.
- */
-void snd_hda_power_up(struct hda_codec *codec)
+/* Transition to powered up, if wait_power_down then wait for a pending
+ * transition to D3 to complete. A pending D3 transition is indicated
+ * with power_transition == -1. */
+static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 {
 	struct hda_bus *bus = codec->bus;
 
 	spin_lock(&codec->power_lock);
 	codec->power_count++;
-	if (codec->power_on || codec->power_transition > 0) {
+	/* Return if power_on or transitioning to power_on, unless currently
+	 * powering down. */
+	if ((codec->power_on || codec->power_transition > 0) &&
+	    !(wait_power_down && codec->power_transition < 0)) {
 		spin_unlock(&codec->power_lock);
 		return;
 	}
@@ -4430,8 +4429,37 @@ void snd_hda_power_up(struct hda_codec *codec)
 	codec->power_transition = 0;
 	spin_unlock(&codec->power_lock);
 }
+
+/**
+ * snd_hda_power_up - Power-up the codec
+ * @codec: HD-audio codec
+ *
+ * Increment the power-up counter and power up the hardware really when
+ * not turned on yet.
+ */
+void snd_hda_power_up(struct hda_codec *codec)
+{
+	__snd_hda_power_up(codec, false);
+}
 EXPORT_SYMBOL_HDA(snd_hda_power_up);
 
+/**
+ * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending
+ *   D3 transition to complete.  This differs from snd_hda_power_up() when
+ *   power_transition == -1.  snd_hda_power_up sees this case as a nop,
+ *   snd_hda_power_up_d3wait waits for the D3 transition to complete then powers
+ *   back up.
+ * @codec: HD-audio codec
+ *
+ * Cancel any power down operation hapenning on the work queue, then power up.
+ */
+void snd_hda_power_up_d3wait(struct hda_codec *codec)
+{
+	/* This will cancel and wait for pending power_work to complete. */
+	__snd_hda_power_up(codec, true);
+}
+EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait);
+
 #define power_save(codec)	\
 	((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
 

+ 2 - 0
sound/pci/hda/hda_codec.h

@@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg);
  */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_up_d3wait(struct hda_codec *codec);
 void snd_hda_power_down(struct hda_codec *codec);
 void snd_hda_update_power_acct(struct hda_codec *codec);
 #else
 static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {}
 static inline void snd_hda_power_down(struct hda_codec *codec) {}
 #endif
 

+ 1 - 1
sound/pci/hda/hda_intel.c

@@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				   buff_step);
 	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
 				   buff_step);
-	snd_hda_power_up(apcm->codec);
+	snd_hda_power_up_d3wait(apcm->codec);
 	err = hinfo->ops.open(hinfo, apcm->codec, substream);
 	if (err < 0) {
 		azx_release_device(azx_dev);

+ 12 - 14
sound/soc/codecs/wm8904.c

@@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 				return ret;
 			}
 
+			regcache_cache_only(wm8904->regmap, false);
 			regcache_sync(wm8904->regmap);
 
 			/* Enable bias */
@@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
 				    WM8904_BIAS_ENA, 0);
 
-#ifdef CONFIG_REGULATOR
-		/* Post 2.6.34 we will be able to get a callback when
-		 * the regulators are disabled which we can use but
-		 * for now just assume that the power will be cut if
-		 * the regulator API is in use.
-		 */
-		codec->cache_sync = 1;
-#endif
+		regcache_cache_only(wm8904->regmap, true);
+		regcache_mark_dirty(wm8904->regmap);
 
 		regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
 				       wm8904->supplies);
@@ -2084,10 +2079,8 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	struct wm8904_pdata *pdata = wm8904->pdata;
-	u16 *reg_cache = codec->reg_cache;
 	int ret, i;
 
-	codec->cache_sync = 1;
 	codec->control_data = wm8904->regmap;
 
 	switch (wm8904->devtype) {
@@ -2150,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 		goto err_enable;
 	}
 
+	regcache_cache_only(wm8904->regmap, true);
 	/* Change some default settings - latch VU and enable ZC */
 	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
 			    WM8904_ADC_VU, WM8904_ADC_VU);
@@ -2180,14 +2174,18 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 			if (!pdata->gpio_cfg[i])
 				continue;
 
-			reg_cache[WM8904_GPIO_CONTROL_1 + i]
-				= pdata->gpio_cfg[i] & 0xffff;
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_GPIO_CONTROL_1 + i,
+					   0xffff,
+					   pdata->gpio_cfg[i]);
 		}
 
 		/* Zero is the default value for these anyway */
 		for (i = 0; i < WM8904_MIC_REGS; i++)
-			reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i]
-				= pdata->mic_cfg[i];
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_MIC_BIAS_CONTROL_0 + i,
+					   0xffff,
+					   pdata->mic_cfg[i]);
 	}
 
 	/* Set Class W by default - this will be managed by the Class

+ 3 - 5
sound/soc/codecs/wm8996.c

@@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 		}
 	}
 
-	regcache_cache_only(codec->control_data, true);
-
 	/* Apply platform data settings */
 	snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
 			    WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
@@ -3051,7 +3049,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
 	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
 		regulator_unregister_notifier(wm8996->supplies[i].consumer,
 					      &wm8996->disable_nb[i]);
-	regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
 
 	return 0;
 }
@@ -3206,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 	dev_info(&i2c->dev, "revision %c\n",
 		 (reg & WM8996_CHIP_REV_MASK) + 'A');
 
-	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
-
 	ret = wm8996_reset(wm8996);
 	if (ret < 0) {
 		dev_err(&i2c->dev, "Failed to issue reset\n");
 		goto err_regmap;
 	}
 
+	regcache_cache_only(wm8996->regmap, true);
+	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+
 	wm8996_init_gpio(wm8996);
 
 	ret = snd_soc_register_codec(&i2c->dev,

+ 11 - 27
sound/soc/pxa/pxa-ssp.c

@@ -33,7 +33,6 @@
 
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <mach/audio.h>
 
 #include "../../arm/pxa2xx-pcm.h"
 #include "pxa-ssp.h"
@@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
 {
 	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 
-	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+	if (ssp->type == PXA25x_SSP) {
 		sscr0 &= ~0x0000ff00;
 		sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
 	} else {
@@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
 	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 	u32 div;
 
-	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+	if (ssp->type == PXA25x_SSP)
 		div = ((sscr0 >> 8) & 0xff) * 2 + 2;
 	else
 		div = ((sscr0 >> 8) & 0xfff) + 1;
@@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		break;
 	case PXA_SSP_CLK_PLL:
 		/* Internal PLL is fixed */
-		if (cpu_is_pxa25x())
+		if (ssp->type == PXA25x_SSP)
 			priv->sysclk = 1843200;
 		else
 			priv->sysclk = 13000000;
@@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 	/* The SSP clock must be disabled when changing SSP clock mode
 	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
-	if (!cpu_is_pxa3xx())
+	if (ssp->type != PXA3xx_SSP)
 		clk_disable(ssp->clk);
 	val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
 	pxa_ssp_write_reg(ssp, SSCR0, val);
-	if (!cpu_is_pxa3xx())
+	if (ssp->type != PXA3xx_SSP)
 		clk_enable(ssp->clk);
 
 	return 0;
@@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 	case PXA_SSP_AUDIO_DIV_SCDB:
 		val = pxa_ssp_read_reg(ssp, SSACD);
 		val &= ~SSACD_SCDB;
-#if defined(CONFIG_PXA3xx)
-		if (cpu_is_pxa3xx())
+		if (ssp->type == PXA3xx_SSP)
 			val &= ~SSACD_SCDX8;
-#endif
 		switch (div) {
 		case PXA_SSP_CLK_SCDB_1:
 			val |= SSACD_SCDB;
 			break;
 		case PXA_SSP_CLK_SCDB_4:
 			break;
-#if defined(CONFIG_PXA3xx)
 		case PXA_SSP_CLK_SCDB_8:
-			if (cpu_is_pxa3xx())
+			if (ssp->type == PXA3xx_SSP)
 				val |= SSACD_SCDX8;
 			else
 				return -EINVAL;
 			break;
-#endif
 		default:
 			return -EINVAL;
 		}
@@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 	struct ssp_device *ssp = priv->ssp;
 	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
 
-#if defined(CONFIG_PXA3xx)
-	if (cpu_is_pxa3xx())
+	if (ssp->type == PXA3xx_SSP)
 		pxa_ssp_write_reg(ssp, SSACDD, 0);
-#endif
 
 	switch (freq_out) {
 	case 5622000:
@@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 		break;
 
 	default:
-#ifdef CONFIG_PXA3xx
 		/* PXA3xx has a clock ditherer which can be used to generate
 		 * a wider range of frequencies - calculate a value for it.
 		 */
-		if (cpu_is_pxa3xx()) {
+		if (ssp->type == PXA3xx_SSP) {
 			u32 val;
 			u64 tmp = 19968;
 			tmp *= 1000000;
@@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 				val, freq_out);
 			break;
 		}
-#endif
 
 		return -EINVAL;
 	}
@@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	/* bit size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-#ifdef CONFIG_PXA3xx
-		if (cpu_is_pxa3xx())
+		if (ssp->type == PXA3xx_SSP)
 			sscr0 |= SSCR0_FPCKE;
-#endif
 		sscr0 |= SSCR0_DataSize(16);
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
@@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			* trying and failing a lot; some of the registers
 			* needed for that mode are only available on PXA3xx.
 			*/
-
-#ifdef CONFIG_PXA3xx
-			if (!cpu_is_pxa3xx())
+			if (ssp->type != PXA3xx_SSP)
 				return -EINVAL;
 
 			sspsp |= SSPSP_SFRMWDTH(width * 2);
@@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			sspsp |= SSPSP_EDMYSTOP(3);
 			sspsp |= SSPSP_DMYSTOP(3);
 			sspsp |= SSPSP_DMYSTRT(1);
-#else
-			return -EINVAL;
-#endif
 		} else {
 			/* The frame width is the width the LRCLK is
 			 * asserted for; the delay is expressed in

+ 13 - 0
sound/soc/tegra/tegra_wm8903.c

@@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+static int tegra_wm8903_remove(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]);
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	wm8903_mic_detect(codec, NULL, 0, 0);
+
+	return 0;
+}
+
 static struct snd_soc_dai_link tegra_wm8903_dai = {
 	.name = "WM8903",
 	.stream_name = "WM8903 PCM",
@@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
 	.dai_link = &tegra_wm8903_dai,
 	.num_links = 1,
 
+	.remove = tegra_wm8903_remove,
+
 	.controls = tegra_wm8903_controls,
 	.num_controls = ARRAY_SIZE(tegra_wm8903_controls),
 	.dapm_widgets = tegra_wm8903_dapm_widgets,

+ 1 - 1
sound/usb/6fire/firmware.c

@@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload(
 	int ret;
 	u8 data;
 	struct usb_device *device = interface_to_usbdev(intf);
-	const struct firmware *fw = 0;
+	const struct firmware *fw = NULL;
 	struct ihex_record *rec = kmalloc(sizeof(struct ihex_record),
 			GFP_KERNEL);
 

+ 13 - 8
sound/usb/pcm.c

@@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 		    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
 		     get_endpoint(alts, 1)->bSynchAddress != 0 &&
 		     !implicit_fb)) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
-				   dev->devnum, fmt->iface, fmt->altsetting);
+			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+				   dev->devnum, fmt->iface, fmt->altsetting,
+				   get_endpoint(alts, 1)->bmAttributes,
+				   get_endpoint(alts, 1)->bLength,
+				   get_endpoint(alts, 1)->bSynchAddress);
 			return -EINVAL;
 		}
 		ep = get_endpoint(alts, 1)->bEndpointAddress;
-		if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+		if (!implicit_fb &&
+		    get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
 		    (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
-		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) ||
-		     ( is_playback && !implicit_fb))) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
-				   dev->devnum, fmt->iface, fmt->altsetting);
+		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+				   dev->devnum, fmt->iface, fmt->altsetting,
+				   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
 			return -EINVAL;
 		}
 
@@ -1147,7 +1151,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
 	return -EINVAL;
 }
 
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream,
+					     int cmd)
 {
 	int err;
 	struct snd_usb_substream *subs = substream->runtime->private_data;