From 6a40dc5ab5036722d8102ba7190dbd9d72982637 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Mon, 13 Oct 2014 11:37:18 +0530
Subject: ALSA: au88x0: added reference of vortex_t

added a pointer of the vortex in the following functions :

vortex_alsafmt_aspfmt
vortex_Vort3D_InitializeSource
a3dsrc_ZeroStateA3D

so that we can have a reference of the vortex in the function.
this reference of the vortex will actually be used in a later patch
to convert the pr_* macro to dev_*.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0.h      | 4 ++--
 sound/pci/au88x0/au88x0_a3d.c  | 6 +++---
 sound/pci/au88x0/au88x0_core.c | 5 +++--
 sound/pci/au88x0/au88x0_pcm.c  | 2 +-
 4 files changed, 9 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index 466a5c8e8354..3a8fefefea77 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -243,7 +243,7 @@ static int vortex_core_init(vortex_t * card);
 static int vortex_core_shutdown(vortex_t * card);
 static void vortex_enable_int(vortex_t * card);
 static irqreturn_t vortex_interrupt(int irq, void *dev_id);
-static int vortex_alsafmt_aspfmt(int alsafmt);
+static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v);
 
 /* Connection  stuff. */
 static void vortex_connect_default(vortex_t * vortex, int en);
@@ -278,7 +278,7 @@ static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix,
 static void vortex_Vort3D_enable(vortex_t * v);
 static void vortex_Vort3D_disable(vortex_t * v);
 static void vortex_Vort3D_connect(vortex_t * vortex, int en);
-static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en);
+static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v);
 #endif
 
 /* Driver stuff. */
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index 30f760e3d2c0..bc9cda3aa725 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -484,7 +484,7 @@ static void a3dsrc_ZeroState(a3dsrc_t * a)
 }
 
 /* Reset entire A3D engine */
-static void a3dsrc_ZeroStateA3D(a3dsrc_t * a)
+static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v)
 {
 	int i, var, var2;
 
@@ -601,7 +601,7 @@ static void vortex_Vort3D_enable(vortex_t *v)
 	Vort3DRend_Initialize(v, XT_HEADPHONE);
 	for (i = 0; i < NR_A3D; i++) {
 		vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2);
-		a3dsrc_ZeroStateA3D(&(v->a3d[0]));
+		a3dsrc_ZeroStateA3D(&v->a3d[0], v);
 	}
 	/* Register ALSA controls */
 	vortex_a3d_register_controls(v);
@@ -676,7 +676,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en)
 }
 
 /* Initialize one single A3D source. */
-static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en)
+static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v)
 {
 	if (a->vortex == NULL) {
 		pr_warn
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 72e81286b70e..00e209617c52 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2177,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 				return -EBUSY;
 			}
 			/* (De)Initialize A3D hardware source. */
-			vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en);
+			vortex_Vort3D_InitializeSource(&vortex->a3d[a3d], en,
+						       vortex);
 		}
 		/* Make SPDIF out exclusive to "spdif" device when in use. */
 		if ((stream->type == VORTEX_PCM_SPDIF) && (en)) {
@@ -2765,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex)
 
 /* Alsa support. */
 
-static int vortex_alsafmt_aspfmt(int alsafmt)
+static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v)
 {
 	int fmt;
 
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 5adc6b92ffab..bdde182f1372 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -332,7 +332,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream)
 		dir = 1;
 	else
 		dir = 0;
-	fmt = vortex_alsafmt_aspfmt(runtime->format);
+	fmt = vortex_alsafmt_aspfmt(runtime->format, chip);
 	spin_lock_irq(&chip->lock);
 	if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
 		vortex_adbdma_setmode(chip, dma, 1, dir, fmt,
-- 
cgit v1.2.3


From 70c84418bf74f582e29906f1eeb19f2e9da53ddd Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Mon, 13 Oct 2014 11:37:19 +0530
Subject: ALSA: au88x0: pr_* replaced with dev_*

pr_* macros replaced with dev_* as they are more preffered over pr_*.
each file which had pr_* was reviewed manually and replaced with dev_*.
here we have actually used the reference of the vortex which was added
to some functions in the previous patch of this series.

The prefix of the CARD_NAME and prefix of "vortex:" was also
removed as the dev_* will print the device name.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0.c        | 33 +++++++-------
 sound/pci/au88x0/au88x0_a3d.c    | 15 ++++---
 sound/pci/au88x0/au88x0_core.c   | 97 +++++++++++++++++++++-------------------
 sound/pci/au88x0/au88x0_eq.c     |  3 +-
 sound/pci/au88x0/au88x0_game.c   |  3 +-
 sound/pci/au88x0/au88x0_mpu401.c |  2 +-
 sound/pci/au88x0/au88x0_pcm.c    |  6 +--
 sound/pci/au88x0/au88x0_synth.c  | 17 ++++---
 8 files changed, 94 insertions(+), 82 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 21ce31f636bc..e9c3833f6d44 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -48,11 +48,10 @@ static void vortex_fix_latency(struct pci_dev *vortex)
 {
 	int rc;
 	if (!(rc = pci_write_config_byte(vortex, 0x40, 0xff))) {
-			pr_info( CARD_NAME
-			       ": vortex latency is 0xff\n");
+			dev_info(&vortex->dev, "vortex latency is 0xff\n");
 	} else {
-		pr_warn( CARD_NAME
-				": could not set vortex latency: pci error 0x%x\n", rc);
+		dev_warn(&vortex->dev,
+			 "could not set vortex latency: pci error 0x%x\n", rc);
 	}
 }
 
@@ -70,11 +69,10 @@ static void vortex_fix_agp_bridge(struct pci_dev *via)
 	if (!(rc = pci_read_config_byte(via, 0x42, &value))
 			&& ((value & 0x10)
 				|| !(rc = pci_write_config_byte(via, 0x42, value | 0x10)))) {
-		pr_info( CARD_NAME
-				": bridge config is 0x%x\n", value | 0x10);
+		dev_info(&via->dev, "bridge config is 0x%x\n", value | 0x10);
 	} else {
-		pr_warn( CARD_NAME
-				": could not set vortex latency: pci error 0x%x\n", rc);
+		dev_warn(&via->dev,
+			 "could not set vortex latency: pci error 0x%x\n", rc);
 	}
 }
 
@@ -97,7 +95,8 @@ static void snd_vortex_workaround(struct pci_dev *vortex, int fix)
 					PCI_DEVICE_ID_AMD_FE_GATE_7007, NULL);
 		}
 		if (via) {
-			pr_info( CARD_NAME ": Activating latency workaround...\n");
+			dev_info(&vortex->dev,
+				 "Activating latency workaround...\n");
 			vortex_fix_latency(vortex);
 			vortex_fix_agp_bridge(via);
 		}
@@ -153,7 +152,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
 		return err;
 	if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 ||
 	    pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) {
-		pr_err( "error to set DMA mask\n");
+		dev_err(card->dev, "error to set DMA mask\n");
 		pci_disable_device(pci);
 		return -ENXIO;
 	}
@@ -182,7 +181,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
 
 	chip->mmio = pci_ioremap_bar(pci, 0);
 	if (!chip->mmio) {
-		pr_err( "MMIO area remap failed.\n");
+		dev_err(card->dev, "MMIO area remap failed.\n");
 		err = -ENOMEM;
 		goto ioremap_out;
 	}
@@ -191,14 +190,14 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
 	 * This must be done before we do request_irq otherwise we can get spurious
 	 * interrupts that we do not handle properly and make a mess of things */
 	if ((err = vortex_core_init(chip)) != 0) {
-		pr_err( "hw core init failed\n");
+		dev_err(card->dev, "hw core init failed\n");
 		goto core_out;
 	}
 
 	if ((err = request_irq(pci->irq, vortex_interrupt,
 			       IRQF_SHARED, KBUILD_MODNAME,
 	                       chip)) != 0) {
-		pr_err( "cannot grab irq\n");
+		dev_err(card->dev, "cannot grab irq\n");
 		goto irq_out;
 	}
 	chip->irq = pci->irq;
@@ -342,11 +341,11 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	chip->rev = pci->revision;
 #ifdef CHIP_AU8830
 	if ((chip->rev) != 0xfe && (chip->rev) != 0xfa) {
-		pr_alert(
-		       "vortex: The revision (%x) of your card has not been seen before.\n",
+		dev_alert(card->dev,
+			  "The revision (%x) of your card has not been seen before.\n",
 		       chip->rev);
-		pr_alert(
-		       "vortex: Please email the results of 'lspci -vv' to openvortex-dev@nongnu.org.\n");
+		dev_alert(card->dev,
+			  "Please email the results of 'lspci -vv' to openvortex-dev@nongnu.org.\n");
 		snd_card_free(card);
 		err = -ENODEV;
 		return err;
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index bc9cda3aa725..ab0f87312911 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -489,7 +489,8 @@ static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v)
 	int i, var, var2;
 
 	if ((a->vortex) == NULL) {
-		pr_err( "vortex: ZeroStateA3D: ERROR: a->vortex is NULL\n");
+		dev_err(v->card->dev,
+			"ZeroStateA3D: ERROR: a->vortex is NULL\n");
 		return;
 	}
 
@@ -628,15 +629,15 @@ static void vortex_Vort3D_connect(vortex_t * v, int en)
 	v->mixxtlk[0] =
 	    vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN);
 	if (v->mixxtlk[0] < 0) {
-		pr_warn
-		    ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n");
+		dev_warn(v->card->dev,
+			 "vortex_Vort3D: ERROR: not enough free mixer resources.\n");
 		return;
 	}
 	v->mixxtlk[1] =
 	    vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN);
 	if (v->mixxtlk[1] < 0) {
-		pr_warn
-		    ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n");
+		dev_warn(v->card->dev,
+			 "vortex_Vort3D: ERROR: not enough free mixer resources.\n");
 		return;
 	}
 #endif
@@ -679,8 +680,8 @@ static void vortex_Vort3D_connect(vortex_t * v, int en)
 static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v)
 {
 	if (a->vortex == NULL) {
-		pr_warn
-		    ("vortex: Vort3D_InitializeSource: A3D source not initialized\n");
+		dev_warn(v->card->dev,
+			 "Vort3D_InitializeSource: A3D source not initialized\n");
 		return;
 	}
 	if (en) {
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 00e209617c52..4667c3232b7f 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -285,8 +285,8 @@ vortex_mixer_addWTD(vortex_t * vortex, unsigned char mix, unsigned char ch)
 		temp = hwread(vortex->mmio, prev);
 		//printk(KERN_INFO "vortex: mixAddWTD: while addr=%x, val=%x\n", prev, temp);
 		if ((++lifeboat) > 0xf) {
-			pr_err(
-			       "vortex_mixer_addWTD: lifeboat overflow\n");
+			dev_err(vortex->card->dev,
+				"vortex_mixer_addWTD: lifeboat overflow\n");
 			return 0;
 		}
 	}
@@ -303,7 +303,7 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch)
 
 	eax = hwread(vortex->mmio, VORTEX_MIXER_SR);
 	if (((1 << ch) & eax) == 0) {
-		pr_err( "mix ALARM %x\n", eax);
+		dev_err(vortex->card->dev, "mix ALARM %x\n", eax);
 		return 0;
 	}
 	ebp = VORTEX_MIXER_CHNBASE + (ch << 2);
@@ -324,8 +324,8 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch)
 			//printk(KERN_INFO "vortex: mixdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src);
 			while ((edx & 0xf) != mix) {
 				if ((esi) > 0xf) {
-					pr_err(
-					       "vortex: mixdelWTD: error lifeboat overflow\n");
+					dev_err(vortex->card->dev,
+						"mixdelWTD: error lifeboat overflow\n");
 					return 0;
 				}
 				esp14 = ebx;
@@ -492,7 +492,7 @@ vortex_src_persist_convratio(vortex_t * vortex, unsigned char src, int ratio)
 		hwwrite(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2), ratio);
 		temp = hwread(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2));
 		if ((++lifeboat) > 0x9) {
-			pr_err( "Vortex: Src cvr fail\n");
+			dev_err(vortex->card->dev, "Src cvr fail\n");
 			break;
 		}
 	}
@@ -684,8 +684,8 @@ vortex_src_addWTD(vortex_t * vortex, unsigned char src, unsigned char ch)
 		temp = hwread(vortex->mmio, prev);
 		//printk(KERN_INFO "vortex: srcAddWTD: while addr=%x, val=%x\n", prev, temp);
 		if ((++lifeboat) > 0xf) {
-			pr_err(
-			       "vortex_src_addWTD: lifeboat overflow\n");
+			dev_err(vortex->card->dev,
+				"vortex_src_addWTD: lifeboat overflow\n");
 			return 0;
 		}
 	}
@@ -703,7 +703,7 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch)
 
 	eax = hwread(vortex->mmio, VORTEX_SRCBLOCK_SR);
 	if (((1 << ch) & eax) == 0) {
-		pr_err( "src alarm\n");
+		dev_err(vortex->card->dev, "src alarm\n");
 		return 0;
 	}
 	ebp = VORTEX_SRC_CHNBASE + (ch << 2);
@@ -724,8 +724,8 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch)
 			//printk(KERN_INFO "vortex: srcdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src);
 			while ((edx & 0xf) != src) {
 				if ((esi) > 0xf) {
-					pr_warn
-					    ("vortex: srcdelWTD: error, lifeboat overflow\n");
+					dev_warn(vortex->card->dev,
+						 "srcdelWTD: error, lifeboat overflow\n");
 					return 0;
 				}
 				esp14 = ebx;
@@ -819,8 +819,8 @@ vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int stereo, int priority,
 	do {
 		temp = hwread(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2));
 		if (lifeboat++ > 0xbb8) {
-			pr_err(
-			       "Vortex: vortex_fifo_setadbctrl fail\n");
+			dev_err(vortex->card->dev,
+				"vortex_fifo_setadbctrl fail\n");
 			break;
 		}
 	}
@@ -915,7 +915,8 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority,
 	do {
 		temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2));
 		if (lifeboat++ > 0xbb8) {
-			pr_err( "Vortex: vortex_fifo_setwtctrl fail\n");
+			dev_err(vortex->card->dev,
+				"vortex_fifo_setwtctrl fail\n");
 			break;
 		}
 	}
@@ -1042,7 +1043,7 @@ static void vortex_fifo_init(vortex_t * vortex)
 	for (x = NR_ADB - 1; x >= 0; x--) {
 		hwwrite(vortex->mmio, addr, (FIFO_U0 | FIFO_U1));
 		if (hwread(vortex->mmio, addr) != (FIFO_U0 | FIFO_U1))
-			pr_err( "bad adb fifo reset!");
+			dev_err(vortex->card->dev, "bad adb fifo reset!");
 		vortex_fifo_clearadbdata(vortex, x, FIFO_SIZE);
 		addr -= 4;
 	}
@@ -1053,9 +1054,9 @@ static void vortex_fifo_init(vortex_t * vortex)
 	for (x = NR_WT - 1; x >= 0; x--) {
 		hwwrite(vortex->mmio, addr, FIFO_U0);
 		if (hwread(vortex->mmio, addr) != FIFO_U0)
-			pr_err(
-			       "bad wt fifo reset (0x%08x, 0x%08x)!\n",
-			       addr, hwread(vortex->mmio, addr));
+			dev_err(vortex->card->dev,
+				"bad wt fifo reset (0x%08x, 0x%08x)!\n",
+				addr, hwread(vortex->mmio, addr));
 		vortex_fifo_clearwtdata(vortex, x, FIFO_SIZE);
 		addr -= 4;
 	}
@@ -1213,8 +1214,9 @@ static int vortex_adbdma_bufshift(vortex_t * vortex, int adbdma)
 	if (dma->period_virt >= dma->nr_periods)
 		dma->period_virt -= dma->nr_periods;
 	if (delta != 1)
-		pr_info( "vortex: %d virt=%d, real=%d, delta=%d\n",
-		       adbdma, dma->period_virt, dma->period_real, delta);
+		dev_info(vortex->card->dev,
+			 "%d virt=%d, real=%d, delta=%d\n",
+			 adbdma, dma->period_virt, dma->period_real, delta);
 
 	return delta;
 }
@@ -1482,8 +1484,8 @@ static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma)
 	dma->period_real = page;
 
 	if (delta != 1)
-		pr_warn( "vortex: wt virt = %d, delta = %d\n",
-		       dma->period_virt, delta);
+		dev_warn(vortex->card->dev, "wt virt = %d, delta = %d\n",
+			 dma->period_virt, delta);
 
 	return delta;
 }
@@ -1667,9 +1669,9 @@ vortex_adb_addroutes(vortex_t * vortex, unsigned char channel,
 		    hwread(vortex->mmio,
 			   VORTEX_ADB_RTBASE + (temp << 2)) & ADB_MASK;
 		if ((lifeboat++) > ADB_MASK) {
-			pr_err(
-			       "vortex_adb_addroutes: unending route! 0x%x\n",
-			       *route);
+			dev_err(vortex->card->dev,
+				"vortex_adb_addroutes: unending route! 0x%x\n",
+				*route);
 			return;
 		}
 	}
@@ -1703,9 +1705,9 @@ vortex_adb_delroutes(vortex_t * vortex, unsigned char channel,
 		    hwread(vortex->mmio,
 			   VORTEX_ADB_RTBASE + (prev << 2)) & ADB_MASK;
 		if (((lifeboat++) > ADB_MASK) || (temp == ADB_MASK)) {
-			pr_err(
-			       "vortex_adb_delroutes: route not found! 0x%x\n",
-			       route0);
+			dev_err(vortex->card->dev,
+				"vortex_adb_delroutes: route not found! 0x%x\n",
+				route0);
 			return;
 		}
 	}
@@ -2045,7 +2047,9 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
 			}
 		}
 	}
-	pr_err( "vortex: FATAL: ResManager: resource type %d exhausted.\n", restype);
+	dev_err(vortex->card->dev,
+		"FATAL: ResManager: resource type %d exhausted.\n",
+		restype);
 	return -ENOMEM;
 }
 
@@ -2173,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 				memset(stream->resources, 0,
 				       sizeof(unsigned char) *
 				       VORTEX_RESOURCE_LAST);
-				pr_err( "vortex: out of A3D sources. Sorry\n");
+				dev_err(vortex->card->dev,
+					"out of A3D sources. Sorry\n");
 				return -EBUSY;
 			}
 			/* (De)Initialize A3D hardware source. */
@@ -2422,7 +2427,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id)
 	hwread(vortex->mmio, VORTEX_IRQ_SOURCE);
 	// Is at least one IRQ flag set?
 	if (source == 0) {
-		pr_err( "vortex: missing irq source\n");
+		dev_err(vortex->card->dev, "missing irq source\n");
 		return IRQ_NONE;
 	}
 
@@ -2430,19 +2435,19 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id)
 	// Attend every interrupt source.
 	if (unlikely(source & IRQ_ERR_MASK)) {
 		if (source & IRQ_FATAL) {
-			pr_err( "vortex: IRQ fatal error\n");
+			dev_err(vortex->card->dev, "IRQ fatal error\n");
 		}
 		if (source & IRQ_PARITY) {
-			pr_err( "vortex: IRQ parity error\n");
+			dev_err(vortex->card->dev, "IRQ parity error\n");
 		}
 		if (source & IRQ_REG) {
-			pr_err( "vortex: IRQ reg error\n");
+			dev_err(vortex->card->dev, "IRQ reg error\n");
 		}
 		if (source & IRQ_FIFO) {
-			pr_err( "vortex: IRQ fifo error\n");
+			dev_err(vortex->card->dev, "IRQ fifo error\n");
 		}
 		if (source & IRQ_DMA) {
-			pr_err( "vortex: IRQ dma error\n");
+			dev_err(vortex->card->dev, "IRQ dma error\n");
 		}
 		handled = 1;
 	}
@@ -2490,7 +2495,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id)
 	}
 
 	if (!handled) {
-		pr_err( "vortex: unknown irq source %x\n", source);
+		dev_err(vortex->card->dev, "unknown irq source %x\n", source);
 	}
 	return IRQ_RETVAL(handled);
 }
@@ -2547,7 +2552,7 @@ vortex_codec_write(struct snd_ac97 * codec, unsigned short addr, unsigned short
 	while (!(hwread(card->mmio, VORTEX_CODEC_CTRL) & 0x100)) {
 		udelay(100);
 		if (lifeboat++ > POLL_COUNT) {
-			pr_err( "vortex: ac97 codec stuck busy\n");
+			dev_err(card->card->dev, "ac97 codec stuck busy\n");
 			return;
 		}
 	}
@@ -2573,7 +2578,7 @@ static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short
 	while (!(hwread(card->mmio, VORTEX_CODEC_CTRL) & 0x100)) {
 		udelay(100);
 		if (lifeboat++ > POLL_COUNT) {
-			pr_err( "vortex: ac97 codec stuck busy\n");
+			dev_err(card->card->dev, "ac97 codec stuck busy\n");
 			return 0xffff;
 		}
 	}
@@ -2587,7 +2592,8 @@ static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short
 		udelay(100);
 		data = hwread(card->mmio, VORTEX_CODEC_IO);
 		if (lifeboat++ > POLL_COUNT) {
-			pr_err( "vortex: ac97 address never arrived\n");
+			dev_err(card->card->dev,
+				"ac97 address never arrived\n");
 			return 0xffff;
 		}
 	} while ((data & VORTEX_CODEC_ADDMASK) !=
@@ -2684,7 +2690,7 @@ static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode)
 static int vortex_core_init(vortex_t *vortex)
 {
 
-	pr_info( "Vortex: init.... ");
+	dev_info(vortex->card->dev, "init started\n");
 	/* Hardware Init. */
 	hwwrite(vortex->mmio, VORTEX_CTRL, 0xffffffff);
 	msleep(5);
@@ -2729,7 +2735,7 @@ static int vortex_core_init(vortex_t *vortex)
 	//vortex_enable_timer_int(vortex);
 	//vortex_disable_timer_int(vortex);
 
-	pr_info( "done.\n");
+	dev_info(vortex->card->dev, "init.... done.\n");
 	spin_lock_init(&vortex->lock);
 
 	return 0;
@@ -2738,7 +2744,7 @@ static int vortex_core_init(vortex_t *vortex)
 static int vortex_core_shutdown(vortex_t * vortex)
 {
 
-	pr_info( "Vortex: shutdown...");
+	dev_info(vortex->card->dev, "shutdown started\n");
 #ifndef CHIP_AU8820
 	vortex_eq_free(vortex);
 	vortex_Vort3D_disable(vortex);
@@ -2760,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex)
 	msleep(5);
 	hwwrite(vortex->mmio, VORTEX_IRQ_SOURCE, 0xffff);
 
-	pr_info( "done.\n");
+	dev_info(vortex->card->dev, "shutdown.... done.\n");
 	return 0;
 }
 
@@ -2794,7 +2800,8 @@ static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v)
 		break;
 	default:
 		fmt = 0x8;
-		pr_err( "vortex: format unsupported %d\n", alsafmt);
+		dev_err(v->card->dev,
+			"format unsupported %d\n", alsafmt);
 		break;
 	}
 	return fmt;
diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c
index 9404ba73eaf6..9585c5c63b96 100644
--- a/sound/pci/au88x0/au88x0_eq.c
+++ b/sound/pci/au88x0/au88x0_eq.c
@@ -845,7 +845,8 @@ snd_vortex_peaks_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *u
 
 	vortex_Eqlzr_GetAllPeaks(vortex, peaks, &count);
 	if (count != 20) {
-		pr_err( "vortex: peak count error 20 != %d \n", count);
+		dev_err(vortex->card->dev,
+			"peak count error 20 != %d\n", count);
 		return -1;
 	}
 	for (i = 0; i < 20; i++)
diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c
index 72daf6cf8169..151815b857a0 100644
--- a/sound/pci/au88x0/au88x0_game.c
+++ b/sound/pci/au88x0/au88x0_game.c
@@ -98,7 +98,8 @@ static int vortex_gameport_register(vortex_t *vortex)
 
 	vortex->gameport = gp = gameport_allocate_port();
 	if (!gp) {
-		pr_err( "vortex: cannot allocate memory for gameport\n");
+		dev_err(vortex->card->dev,
+			"cannot allocate memory for gameport\n");
 		return -ENOMEM;
 	}
 
diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
index 328c1943c0c3..1025e55ca854 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -73,7 +73,7 @@ static int snd_vortex_midi(vortex_t *vortex)
 	/* Check if anything is OK. */
 	temp = hwread(vortex->mmio, VORTEX_MIDI_DATA);
 	if (temp != MPU401_ACK /*0xfe */ ) {
-		pr_err( "midi port doesn't acknowledge!\n");
+		dev_err(vortex->card->dev, "midi port doesn't acknowledge!\n");
 		return -ENODEV;
 	}
 	/* Enable MPU401 interrupts. */
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index bdde182f1372..a6d6d8d0867a 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -227,7 +227,7 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream,
 	err =
 	    snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
 	if (err < 0) {
-		pr_err( "Vortex: pcm page alloc failed!\n");
+		dev_err(chip->card->dev, "Vortex: pcm page alloc failed!\n");
 		return err;
 	}
 	/*
@@ -371,7 +371,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		}
 #ifndef CHIP_AU8810
 		else {
-			pr_info( "vortex: wt start %d\n", dma);
+			dev_info(chip->card->dev, "wt start %d\n", dma);
 			vortex_wtdma_startfifo(chip, dma);
 		}
 #endif
@@ -384,7 +384,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			vortex_adbdma_stopfifo(chip, dma);
 #ifndef CHIP_AU8810
 		else {
-			pr_info( "vortex: wt stop %d\n", dma);
+			dev_info(chip->card->dev, "wt stop %d\n", dma);
 			vortex_wtdma_stopfifo(chip, dma);
 		}
 #endif
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index f094bac24291..78e12f4796f3 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -90,7 +90,7 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch)
 	hwwrite(vortex->mmio, WT_PARM(wt, 2), 0);
 
 	temp = hwread(vortex->mmio, WT_PARM(wt, 3));
-	pr_debug( "vortex: WT PARM3: %x\n", temp);
+	dev_dbg(vortex->card->dev, "WT PARM3: %x\n", temp);
 	//hwwrite(vortex->mmio, WT_PARM(wt, 3), temp);
 
 	hwwrite(vortex->mmio, WT_DELAY(wt, 0), 0);
@@ -98,7 +98,8 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch)
 	hwwrite(vortex->mmio, WT_DELAY(wt, 2), 0);
 	hwwrite(vortex->mmio, WT_DELAY(wt, 3), 0);
 
-	pr_debug( "vortex: WT GMODE: %x\n", hwread(vortex->mmio, WT_GMODE(wt)));
+	dev_dbg(vortex->card->dev, "WT GMODE: %x\n",
+		hwread(vortex->mmio, WT_GMODE(wt)));
 
 	hwwrite(vortex->mmio, WT_PARM(wt, 2), 0xffffffff);
 	hwwrite(vortex->mmio, WT_PARM(wt, 3), 0xcff1c810);
@@ -106,7 +107,8 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch)
 	voice->parm0 = voice->parm1 = 0xcfb23e2f;
 	hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0);
 	hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1);
-	pr_debug( "vortex: WT GMODE 2 : %x\n", hwread(vortex->mmio, WT_GMODE(wt)));
+	dev_dbg(vortex->card->dev, "WT GMODE 2 : %x\n",
+		hwread(vortex->mmio, WT_GMODE(wt)));
 	return 0;
 }
 
@@ -196,14 +198,15 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
 
 	if ((reg == 5) || ((reg >= 7) && (reg <= 10)) || (reg == 0xc)) {
 		if (wt >= (NR_WT / NR_WT_PB)) {
-			pr_warn
-			    ("vortex: WT SetReg: bank out of range. reg=0x%x, wt=%d\n",
-			     reg, wt);
+			dev_warn(vortex->card->dev,
+				 "WT SetReg: bank out of range. reg=0x%x, wt=%d\n",
+				 reg, wt);
 			return 0;
 		}
 	} else {
 		if (wt >= NR_WT) {
-			pr_err( "vortex: WT SetReg: voice out of range\n");
+			dev_err(vortex->card->dev,
+				"WT SetReg: voice out of range\n");
 			return 0;
 		}
 	}
-- 
cgit v1.2.3


From 03ad6a8c93b6df2d65c305b5b5f9474068b45bfb Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 16 Oct 2014 15:33:45 +0200
Subject: ALSA: hda - Fix "PCM" name being used on one DAC when there are two
 DACs

In the scenario where there is one "Line Out", one "Speaker" and one
"Headphone", and there are only two DACs, two outputs will share a DAC.
Currently any mixer on such a DAC will get the "PCM" name, which is
misleading. Instead use "Headphone+LO" or "Speaker+LO" to better
specify what the volume actually controls.

[fixed missing slave string additions by tiwai]

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 15 ++++++++++++++-
 1 file changed, 14 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 64220c08bd98..dc13cce70932 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -1038,6 +1038,19 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
 			break;
 		*index = ch;
 		return "Headphone";
+	case AUTO_PIN_LINE_OUT:
+		/* This deals with the case where we have two DACs and
+		 * one LO, one HP and one Speaker */
+		if (!ch && cfg->speaker_outs && cfg->hp_outs) {
+			bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type);
+			bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type);
+			if (hp_lo_shared && spk_lo_shared)
+				return spec->vmaster_mute.hook ? "PCM" : "Master";
+			if (hp_lo_shared)
+				return "Headphone+LO";
+			if (spk_lo_shared)
+				return "Speaker+LO";
+		}
 	}
 
 	/* for a single channel output, we don't have to name the channel */
@@ -4524,7 +4537,7 @@ static const char * const slave_pfxs[] = {
 	"CLFE", "Bass Speaker", "PCM",
 	"Speaker Front", "Speaker Surround", "Speaker CLFE", "Speaker Side",
 	"Headphone Front", "Headphone Surround", "Headphone CLFE",
-	"Headphone Side",
+	"Headphone Side", "Headphone+LO", "Speaker+LO",
 	NULL,
 };
 
-- 
cgit v1.2.3


From 3abb4f4d0e7aaad0d12004b5057f4486a688752b Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 16 Oct 2014 15:33:46 +0200
Subject: ALSA: hda - Use "Line Out" name instead of "PCM" when there are other
 outputs

In case there are speakers or headphones as well, anything that only
covers the line out should not be labelled "PCM". Let's name it
"Line Out" instead for clarity.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index dc13cce70932..06d721085e72 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -1055,7 +1055,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
 
 	/* for a single channel output, we don't have to name the channel */
 	if (cfg->line_outs == 1 && !spec->multi_ios)
-		return "PCM";
+		return "Line Out";
 
 	if (ch >= ARRAY_SIZE(channel_name)) {
 		snd_BUG();
-- 
cgit v1.2.3


From 3b7a00dc9e4277d6fcad68dd1db35f77264ede5f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:15:36 +0200
Subject: ALSA: ac97: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c |  10 +--
 sound/pci/ac97/ac97_patch.c | 176 +++++++++++++-------------------------------
 2 files changed, 53 insertions(+), 133 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 14ad54b7928c..5ee2f17c287c 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -463,14 +463,8 @@ static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value;
 	
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
-	uinfo->value.enumerated.items = e->mask;
-	
-	if (uinfo->value.enumerated.item > e->mask - 1)
-		uinfo->value.enumerated.item = e->mask - 1;
-	strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2,
+				 e->mask, e->texts);
 }
 
 static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 991762215417..50f420d69a8a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -81,22 +81,11 @@ static int ac97_update_bits_page(struct snd_ac97 *ac97, unsigned short reg, unsi
 /*
  * shared line-in/mic controls
  */
-static int ac97_enum_text_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo,
-			       const char **texts, unsigned int nums)
-{
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = nums;
-	if (uinfo->value.enumerated.item > nums - 1)
-		uinfo->value.enumerated.item = nums - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
 static int ac97_surround_jack_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static const char *texts[] = { "Shared", "Independent" };
-	return ac97_enum_text_info(kcontrol, uinfo, texts, 2);
+	static const char * const texts[] = { "Shared", "Independent" };
+
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int ac97_surround_jack_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -123,9 +112,9 @@ static int ac97_surround_jack_mode_put(struct snd_kcontrol *kcontrol, struct snd
 
 static int ac97_channel_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static const char *texts[] = { "2ch", "4ch", "6ch", "8ch" };
-	return ac97_enum_text_info(kcontrol, uinfo, texts,
-		kcontrol->private_value);
+	static const char * const texts[] = { "2ch", "4ch", "6ch", "8ch" };
+
+	return snd_ctl_enum_info(uinfo, 1, kcontrol->private_value, texts);
 }
 
 static int ac97_channel_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -240,17 +229,11 @@ static inline int alc850_is_aux_back_surround(struct snd_ac97 *ac97)
 static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = {
+	static const char * const texts[3] = {
 		"Standard", "Small", "Smaller"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol,
@@ -293,15 +276,9 @@ static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker =
 static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol,
 					     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2] = { "AC-Link", "A/D Converter" };
+	static const char * const texts[2] = { "AC-Link", "A/D Converter" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol,
@@ -401,15 +378,9 @@ static int patch_yamaha_ymf743(struct snd_ac97 *ac97)
    There is also a bit to mute S/PDIF output in a vendor-specific register. */
 static int snd_ac97_ymf753_spdif_output_pin_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = { "Disabled", "Pin 43", "Pin 48" };
+	static const char * const texts[3] = { "Disabled", "Pin 43", "Pin 48" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_ac97_ymf753_spdif_output_pin_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1103,16 +1074,11 @@ static int patch_sigmatel_stac9756(struct snd_ac97 * ac97)
 
 static int snd_ac97_stac9758_output_jack_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[5] = { "Input/Disabled", "Front Output",
+	static const char * const texts[5] = {
+		"Input/Disabled", "Front Output",
 		"Rear Output", "Center/LFE Output", "Mixer Output" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 5;
-	if (uinfo->value.enumerated.item > 4)
-		uinfo->value.enumerated.item = 4;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 5, texts);
 }
 
 static int snd_ac97_stac9758_output_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1147,16 +1113,11 @@ static int snd_ac97_stac9758_output_jack_put(struct snd_kcontrol *kcontrol, stru
 
 static int snd_ac97_stac9758_input_jack_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[7] = { "Mic2 Jack", "Mic1 Jack", "Line In Jack",
+	static const char * const texts[7] = {
+		"Mic2 Jack", "Mic1 Jack", "Line In Jack",
 		"Front Jack", "Rear Jack", "Center/LFE Jack", "Mute" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 7;
-	if (uinfo->value.enumerated.item > 6)
-		uinfo->value.enumerated.item = 6;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 7, texts);
 }
 
 static int snd_ac97_stac9758_input_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1181,15 +1142,11 @@ static int snd_ac97_stac9758_input_jack_put(struct snd_kcontrol *kcontrol, struc
 
 static int snd_ac97_stac9758_phonesel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = { "None", "Front Jack", "Rear Jack" };
+	static const char * const texts[3] = {
+		"None", "Front Jack", "Rear Jack"
+	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_ac97_stac9758_phonesel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1804,15 +1761,9 @@ static int patch_ad1886(struct snd_ac97 * ac97)
 
 static int snd_ac97_ad198x_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2] = { "AC-Link", "A/D Converter" };
+	static const char * const texts[2] = { "AC-Link", "A/D Converter" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_ac97_ad198x_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1994,15 +1945,9 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd
 
 static int snd_ac97_ad1888_downmix_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = {"Off", "6 -> 4", "6 -> 2"};
+	static const char * const texts[3] = {"Off", "6 -> 4", "6 -> 2"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_ac97_ad1888_downmix_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2153,16 +2098,11 @@ static int patch_ad1980(struct snd_ac97 * ac97)
 static int snd_ac97_ad1985_vrefout_info(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = {"High-Z", "3.7 V", "2.25 V", "0 V"};
+	static const char * const texts[4] = {
+		"High-Z", "3.7 V", "2.25 V", "0 V"
+	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item > 3)
-		uinfo->value.enumerated.item = 3;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 
 static int snd_ac97_ad1985_vrefout_get(struct snd_kcontrol *kcontrol,
@@ -2756,20 +2696,18 @@ static const struct snd_kcontrol_new snd_ac97_controls_alc655[] = {
 
 static int alc655_iec958_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts_655[3] = { "PCM", "Analog In", "IEC958 In" };
-	static char *texts_658[4] = { "PCM", "Analog1 In", "Analog2 In", "IEC958 In" };
+	static const char * const texts_655[3] = {
+		"PCM", "Analog In", "IEC958 In"
+	};
+	static const char * const texts_658[4] = {
+		"PCM", "Analog1 In", "Analog2 In", "IEC958 In"
+	};
 	struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = ac97->spec.dev_flags ? 4 : 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       ac97->spec.dev_flags ?
-	       texts_658[uinfo->value.enumerated.item] :
-	       texts_655[uinfo->value.enumerated.item]);
-	return 0;
+	if (ac97->spec.dev_flags)
+		return snd_ctl_enum_info(uinfo, 1, 4, texts_658);
+	else
+		return snd_ctl_enum_info(uinfo, 1, 3, texts_655);
 }
 
 static int alc655_iec958_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -3055,15 +2993,9 @@ static int patch_cm9738(struct snd_ac97 * ac97)
 
 static int snd_ac97_cmedia_spdif_playback_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Analog", "Digital" };
+	static const char * const texts[] = { "Analog", "Digital" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_ac97_cmedia_spdif_playback_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -3235,15 +3167,9 @@ static const struct snd_kcontrol_new snd_ac97_cm9761_controls[] = {
 
 static int cm9761_spdif_out_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "AC-Link", "ADC", "SPDIF-In" };
+	static const char * const texts[] = { "AC-Link", "ADC", "SPDIF-In" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int cm9761_spdif_out_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -3552,11 +3478,12 @@ static int snd_ac97_vt1617a_smart51_info(struct snd_kcontrol *kcontrol,
 	 * is SM51EN *AND* it's Bit14, not Bit15 so the table is very
 	 * counter-intuitive */ 
 
-	static const char* texts[] = { "LineIn Mic1", "LineIn Mic1 Mic3",
+	static const char * const texts[] = {"LineIn Mic1", "LineIn Mic1 Mic3",
 				       "Surr LFE/C Mic3", "LineIn LFE/C Mic3",
 				       "LineIn Mic2", "LineIn Mic2 Mic1",
 				       "Surr LFE Mic1", "Surr LFE Mic1 Mic2"};
-	return ac97_enum_text_info(kcontrol, uinfo, texts, 8);
+
+	return snd_ctl_enum_info(uinfo, 1, 8, texts);
 }
 
 static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol,
@@ -3720,9 +3647,8 @@ static struct vt1618_uaj_item vt1618_uaj[3] = {
 static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol,
 				    struct snd_ctl_elem_info *uinfo)
 {
-	return ac97_enum_text_info(kcontrol, uinfo,
-				   vt1618_uaj[kcontrol->private_value].items,
-				   4);
+	return snd_ctl_enum_info(uinfo, 1, 4,
+				 vt1618_uaj[kcontrol->private_value].items);
 }
 
 /* All of the vt1618 Universal Audio Jack twiddlers are on
@@ -3767,9 +3693,9 @@ static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol,
 static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static const char *txt_aux[] = {"Aux In", "Back Surr Out"};
+	static const char * const txt_aux[] = {"Aux In", "Back Surr Out"};
 
-	return ac97_enum_text_info(kcontrol, uinfo, txt_aux, 2);
+	return snd_ctl_enum_info(uinfo, 1, 2, txt_aux);
 }
 
 static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 1bc10bb68d348078af0eb8b64292ec542dcd7634 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 21 Oct 2014 08:51:45 +0200
Subject: ALSA: ac97: Constify more text arrays

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_patch.c | 22 ++++++++++++++--------
 sound/pci/ac97/ac97_patch.h |  2 +-
 2 files changed, 15 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 50f420d69a8a..ceaac1c41906 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -33,7 +33,8 @@
 static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
 						    const char *name);
 static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
-				const unsigned int *tlv, const char **slaves);
+				const unsigned int *tlv,
+				const char * const *slaves);
 
 /*
  *  Chip specific initialization
@@ -3196,7 +3197,9 @@ static int cm9761_spdif_out_source_put(struct snd_kcontrol *kcontrol, struct snd
 				    ucontrol->value.enumerated.item[0] == 1 ? 0x2 : 0);
 }
 
-static const char *cm9761_dac_clock[] = { "AC-Link", "SPDIF-In", "Both" };
+static const char * const cm9761_dac_clock[] = {
+	"AC-Link", "SPDIF-In", "Both"
+};
 static const struct ac97_enum cm9761_dac_clock_enum =
 	AC97_ENUM_SINGLE(AC97_CM9761_SPDIF_CTRL, 9, 3, cm9761_dac_clock);
 
@@ -3310,7 +3313,9 @@ static int patch_cm9761(struct snd_ac97 *ac97)
 #define AC97_CM9780_MULTI_CHAN	0x66
 #define AC97_CM9780_SPDIF	0x6c
 
-static const char *cm9780_ch_select[] = { "Front", "Side", "Center/LFE", "Rear" };
+static const char * const cm9780_ch_select[] = {
+	"Front", "Side", "Center/LFE", "Rear"
+};
 static const struct ac97_enum cm9780_ch_select_enum =
 	AC97_ENUM_SINGLE(AC97_CM9780_MULTI_CHAN, 6, 4, cm9780_ch_select);
 static const struct snd_kcontrol_new cm9780_controls[] = {
@@ -3356,7 +3361,7 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0),
 AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0),
 };
 
-static const char *slave_vols_vt1616[] = {
+static const char * const slave_vols_vt1616[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
 	"Center Playback Volume",
@@ -3364,7 +3369,7 @@ static const char *slave_vols_vt1616[] = {
 	NULL
 };
 
-static const char *slave_sws_vt1616[] = {
+static const char * const slave_sws_vt1616[] = {
 	"Front Playback Switch",
 	"Surround Playback Switch",
 	"Center Playback Switch",
@@ -3385,10 +3390,11 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
 
 /* create a virtual master control and add slaves */
 static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
-				const unsigned int *tlv, const char **slaves)
+				const unsigned int *tlv,
+				const char * const *slaves)
 {
 	struct snd_kcontrol *kctl;
-	const char **s;
+	const char * const *s;
 	int err;
 
 	kctl = snd_ctl_make_virtual_master(name, tlv);
@@ -3612,7 +3618,7 @@ static int patch_vt1617a(struct snd_ac97 * ac97)
 struct vt1618_uaj_item {
 	unsigned short mask;
 	unsigned short shift;
-	const char *items[4];
+	const char * const items[4];
 };
 
 /* This list reflects the vt1618 docs for Vendor Defined Register 0x60. */
diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h
index 47bf8dfe8276..d1ce151fe722 100644
--- a/sound/pci/ac97/ac97_patch.h
+++ b/sound/pci/ac97/ac97_patch.h
@@ -49,7 +49,7 @@ struct ac97_enum {
 	unsigned char shift_l;
 	unsigned char shift_r;
 	unsigned short mask;
-	const char **texts;
+	const char * const *texts;
 };
 
 #define AC97_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \
-- 
cgit v1.2.3


From 30d0ae425ab1c9bb0003c3798de78fbf30ddebdc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:15:50 +0200
Subject: ALSA: asihpi: Use snd_ctl_enum_info()

... and reduce the open codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 46 +++++-----------------------------------------
 1 file changed, 5 insertions(+), 41 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 5017176bfaa1..ac66b3228a34 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -1625,18 +1625,7 @@ static const char * const asihpi_aesebu_format_names[] = {
 static int snd_asihpi_aesebu_format_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-
-	strcpy(uinfo->value.enumerated.name,
-		asihpi_aesebu_format_names[uinfo->value.enumerated.item]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, asihpi_aesebu_format_names);
 }
 
 static int snd_asihpi_aesebu_format_get(struct snd_kcontrol *kcontrol,
@@ -1863,22 +1852,7 @@ static int snd_asihpi_tuner_band_info(struct snd_kcontrol *kcontrol,
 	if (num_bands < 0)
 		return num_bands;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = num_bands;
-
-	if (num_bands > 0) {
-		if (uinfo->value.enumerated.item >=
-					uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-
-		strcpy(uinfo->value.enumerated.name,
-			asihpi_tuner_band_names[
-				tuner_bands[uinfo->value.enumerated.item]]);
-
-	}
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, num_bands, asihpi_tuner_band_names);
 }
 
 static int snd_asihpi_tuner_band_get(struct snd_kcontrol *kcontrol,
@@ -2253,7 +2227,7 @@ static int snd_asihpi_cmode_info(struct snd_kcontrol *kcontrol,
 	u32 h_control = kcontrol->private_value;
 	u16 mode;
 	int i;
-	u16 mode_map[6];
+	const char *mapped_names[6];
 	int valid_modes = 0;
 
 	/* HPI channel mode values can be from 1 to 6
@@ -2262,24 +2236,14 @@ static int snd_asihpi_cmode_info(struct snd_kcontrol *kcontrol,
 	for (i = 0; i < HPI_CHANNEL_MODE_LAST; i++)
 		if (!hpi_channel_mode_query_mode(
 			h_control, i, &mode)) {
-			mode_map[valid_modes] = mode;
+			mapped_names[valid_modes] = mode_names[mode];
 			valid_modes++;
 			}
 
 	if (!valid_modes)
 		return -EINVAL;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = valid_modes;
-
-	if (uinfo->value.enumerated.item >= valid_modes)
-		uinfo->value.enumerated.item = valid_modes - 1;
-
-	strcpy(uinfo->value.enumerated.name,
-	       mode_names[mode_map[uinfo->value.enumerated.item]]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, valid_modes, mapped_names);
 }
 
 static int snd_asihpi_cmode_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 4d765e48c5edb2090b82e97680b2d1ddf6d18c31 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:16:01 +0200
Subject: ALSA: aw2: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to the text array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/aw2/aw2-alsa.c | 13 ++-----------
 1 file changed, 2 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3878cf5de9a4..e1cf01949fda 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -725,19 +725,10 @@ static int snd_aw2_new_pcm(struct aw2 *chip)
 static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol,
 					       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2] = {
+	static const char * const texts[2] = {
 		"Analog", "Digital"
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) {
-		uinfo->value.enumerated.item =
-		    uinfo->value.enumerated.items - 1;
-	}
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 9b311a0ad9ec0df9f010bcadd19193b1cee593f6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:16:13 +0200
Subject: ALSA: azt3328: Use snd_ctl_enum_info()

... and reduce the open codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/azt3328.c | 13 ++++---------
 1 file changed, 4 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 5a69e26cb2fb..fdbb9c05c77b 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1034,11 +1034,6 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
 	const char * const *p = NULL;
 
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
-        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-        uinfo->count = (reg.reg == IDX_MIXER_REC_SELECT) ? 2 : 1;
-        uinfo->value.enumerated.items = reg.enum_c;
-        if (uinfo->value.enumerated.item > reg.enum_c - 1U)
-                uinfo->value.enumerated.item = reg.enum_c - 1U;
 	if (reg.reg == IDX_MIXER_ADVCTL2) {
 		switch(reg.lchan_shift) {
 		case 8: /* modem out sel */
@@ -1051,12 +1046,12 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
 			p = texts4;
 			break;
 		}
-	} else
-	if (reg.reg == IDX_MIXER_REC_SELECT)
+	} else if (reg.reg == IDX_MIXER_REC_SELECT)
 		p = texts3;
 
-	strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]);
-        return 0;
+	return snd_ctl_enum_info(uinfo,
+				 (reg.reg == IDX_MIXER_REC_SELECT) ? 2 : 1,
+				 reg.enum_c, p);
 }
 
 static int
-- 
cgit v1.2.3


From de95eae25a2744ba5f9bd3c862bb43a1b177ad58 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:16:26 +0200
Subject: ALSA: ca0106: Use snd_ctl_enum_info()

... and reduce the open codes.  Also correct the array size and add
missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ca0106/ca0106_mixer.c | 40 ++++++++--------------------------------
 1 file changed, 8 insertions(+), 32 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 27de0de90018..68c0eb0a2807 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -185,17 +185,11 @@ static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol,
 static int snd_ca0106_capture_source_info(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[6] = {
+	static const char * const texts[6] = {
 		"IEC958 out", "i2s mixer out", "IEC958 in", "i2s in", "AC97 in", "SRC out"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 6;
-	if (uinfo->value.enumerated.item > 5)
-                uinfo->value.enumerated.item = 5;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 6, texts);
 }
 
 static int snd_ca0106_capture_source_get(struct snd_kcontrol *kcontrol,
@@ -228,17 +222,11 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
 static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[6] = {
+	static const char * const texts[4] = {
 		"Phone", "Mic", "Line in", "Aux"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item > 3)
-                uinfo->value.enumerated.item = 3;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 
 static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
@@ -273,29 +261,17 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
 static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol,
 					       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2] = { "Side out", "Line in" };
+	static const char * const texts[2] = { "Side out", "Line in" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-                uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol,
 					       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2] = { "Line in", "Mic in" };
+	static const char * const texts[2] = { "Line in", "Mic in" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-                uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_ca0106_capture_mic_line_in_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From c69a4f3046ee5a28ab09a1786a73d04bd6177445 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:16:38 +0200
Subject: ALSA: echoaudio: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c | 33 ++++++---------------------------
 1 file changed, 6 insertions(+), 27 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 631aaa4046ad..d82321ff549b 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1416,21 +1416,14 @@ static struct snd_kcontrol_new snd_echo_vmixer = {
 static int snd_echo_digital_mode_info(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_info *uinfo)
 {
-	static char *names[4] = {
+	static const char * const names[4] = {
 		"S/PDIF Coaxial", "S/PDIF Optical", "ADAT Optical",
 		"S/PDIF Cdrom"
 	};
 	struct echoaudio *chip;
 
 	chip = snd_kcontrol_chip(kcontrol);
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->value.enumerated.items = chip->num_digital_modes;
-	uinfo->count = 1;
-	if (uinfo->value.enumerated.item >= chip->num_digital_modes)
-		uinfo->value.enumerated.item = chip->num_digital_modes - 1;
-	strcpy(uinfo->value.enumerated.name, names[
-			chip->digital_mode_list[uinfo->value.enumerated.item]]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, chip->num_digital_modes, names);
 }
 
 static int snd_echo_digital_mode_get(struct snd_kcontrol *kcontrol,
@@ -1509,16 +1502,9 @@ static struct snd_kcontrol_new snd_echo_digital_mode_switch = {
 static int snd_echo_spdif_mode_info(struct snd_kcontrol *kcontrol,
 				    struct snd_ctl_elem_info *uinfo)
 {
-	static char *names[2] = {"Consumer", "Professional"};
+	static const char * const names[2] = {"Consumer", "Professional"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->value.enumerated.items = 2;
-	uinfo->count = 1;
-	if (uinfo->value.enumerated.item)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name,
-	       names[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, names);
 }
 
 static int snd_echo_spdif_mode_get(struct snd_kcontrol *kcontrol,
@@ -1566,21 +1552,14 @@ static struct snd_kcontrol_new snd_echo_spdif_mode_switch = {
 static int snd_echo_clock_source_info(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_info *uinfo)
 {
-	static char *names[8] = {
+	static const char * const names[8] = {
 		"Internal", "Word", "Super", "S/PDIF", "ADAT", "ESync",
 		"ESync96", "MTC"
 	};
 	struct echoaudio *chip;
 
 	chip = snd_kcontrol_chip(kcontrol);
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->value.enumerated.items = chip->num_clock_sources;
-	uinfo->count = 1;
-	if (uinfo->value.enumerated.item >= chip->num_clock_sources)
-		uinfo->value.enumerated.item = chip->num_clock_sources - 1;
-	strcpy(uinfo->value.enumerated.name, names[
-			chip->clock_source_list[uinfo->value.enumerated.item]]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, chip->num_clock_sources, names);
 }
 
 static int snd_echo_clock_source_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 1541c66d3bb78c8a388025b074c75658c790b72f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:16:48 +0200
Subject: ALSA: emu10k1: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/emumixer.c | 58 ++++++++++----------------------------------
 sound/pci/emu10k1/p16v.c     | 20 +++------------
 2 files changed, 17 insertions(+), 61 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index c5ae2a24d8a5..1de33025669a 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -83,7 +83,7 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol,
  * Items labels in enum mixer controls assigning source data to
  * each destination
  */
-static char *emu1010_src_texts[] = { 
+static const char * const emu1010_src_texts[] = {
 	"Silence",
 	"Dock Mic A",
 	"Dock Mic B",
@@ -141,7 +141,7 @@ static char *emu1010_src_texts[] = {
 
 /* 1616(m) cardbus */
 
-static char *emu1616_src_texts[] = {
+static const char * const emu1616_src_texts[] = {
 	"Silence",
 	"Dock Mic A",
 	"Dock Mic B",
@@ -393,23 +393,11 @@ static int snd_emu1010_input_output_source_info(struct snd_kcontrol *kcontrol,
 						struct snd_ctl_elem_info *uinfo)
 {
 	struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
-	char **items;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) {
-		uinfo->value.enumerated.items = 49;
-		items = emu1616_src_texts;
-	} else {
-		uinfo->value.enumerated.items = 53;
-		items = emu1010_src_texts;
-	}
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       items[uinfo->value.enumerated.item]);
-	return 0;
+	if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616)
+		return snd_ctl_enum_info(uinfo, 1, 49, emu1616_src_texts);
+	else
+		return snd_ctl_enum_info(uinfo, 1, 53, emu1010_src_texts);
 }
 
 static int snd_emu1010_output_source_get(struct snd_kcontrol *kcontrol,
@@ -699,19 +687,11 @@ static struct snd_kcontrol_new snd_emu1010_dac_pads[] = {
 static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = {
+	static const char * const texts[4] = {
 		"44100", "48000", "SPDIF", "ADAT"
 	};
 		
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-	
-	
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 
 static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol,
@@ -830,21 +810,15 @@ static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
 #if 0
-	static char *texts[4] = {
+	static const char * const texts[4] = {
 		"Unknown1", "Unknown2", "Mic", "Line"
 	};
 #endif
-	static char *texts[2] = {
+	static const char * const texts[2] = {
 		"Mic", "Line"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-                uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_audigy_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
@@ -997,15 +971,9 @@ static struct snd_kcontrol_new snd_audigy_i2c_volume_ctls[] = {
 #if 0
 static int snd_audigy_spdif_output_rate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"44100", "48000", "96000"};
+	static const char * const texts[] = {"44100", "48000", "96000"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_audigy_spdif_output_rate_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index a4fe7f0c9458..7ef3898a7806 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -757,18 +757,12 @@ static int snd_p16v_volume_put(struct snd_kcontrol *kcontrol,
 static int snd_p16v_capture_source_info(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[8] = {
+	static const char * const texts[8] = {
 		"SPDIF", "I2S", "SRC48", "SRCMulti_SPDIF", "SRCMulti_I2S",
 		"CDIF", "FX", "AC97"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 8;
-	if (uinfo->value.enumerated.item > 7)
-                uinfo->value.enumerated.item = 7;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 8, texts);
 }
 
 static int snd_p16v_capture_source_get(struct snd_kcontrol *kcontrol,
@@ -805,15 +799,9 @@ static int snd_p16v_capture_source_put(struct snd_kcontrol *kcontrol,
 static int snd_p16v_capture_channel_info(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = { "0", "1", "2", "3",  };
+	static const char * const texts[4] = { "0", "1", "2", "3", };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item > 3)
-                uinfo->value.enumerated.item = 3;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 
 static int snd_p16v_capture_channel_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 6b6b295e8053dd5a005aaa089b5bed4b4a65c632 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:17:08 +0200
Subject: ALSA: es1938: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to the text array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/es1938.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 639962443ccc..0fc46eb4e251 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1045,18 +1045,12 @@ static int snd_es1938_new_pcm(struct es1938 *chip, int device)
 static int snd_es1938_info_mux(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[8] = {
+	static const char * const texts[8] = {
 		"Mic", "Mic Master", "CD", "AOUT",
 		"Mic1", "Mix", "Line", "Master"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 8;
-	if (uinfo->value.enumerated.item > 7)
-		uinfo->value.enumerated.item = 7;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 8, texts);
 }
 
 static int snd_es1938_get_mux(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From ca776a28ae10bb06807f23e807f0f459dab78318 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:17:17 +0200
Subject: ALSA: fm801: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to the text array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/fm801.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index c5038303a126..d167afffce5f 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -958,17 +958,11 @@ static int snd_fm801_put_double(struct snd_kcontrol *kcontrol,
 static int snd_fm801_info_mux(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[5] = {
+	static const char * const texts[5] = {
 		"AC97 Primary", "FM", "I2S", "PCM", "AC97 Secondary"
 	};
  
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 5;
-	if (uinfo->value.enumerated.item > 4)
-		uinfo->value.enumerated.item = 4;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 5, texts);
 }
 
 static int snd_fm801_get_mux(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 3ff72219320f616489bf0d98ddac12899da4a9ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:17:28 +0200
Subject: ALSA: hda: Use snd_ctl_enum_info()

... and reduce the open codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 19 ++-----------------
 1 file changed, 2 insertions(+), 17 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 15e0089492f7..259fbeaa37bd 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2927,16 +2927,8 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
 	static const char * const texts[] = {
 		"On", "Off", "Follow Master"
 	};
-	unsigned int index;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	index = uinfo->value.enumerated.item;
-	if (index >= 3)
-		index = 2;
-	strcpy(uinfo->value.enumerated.name, texts[index]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol,
@@ -5195,14 +5187,7 @@ int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
 		texts = texts_default;
 	}
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = num_items;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, num_items, texts);
 }
 EXPORT_SYMBOL_GPL(snd_hda_enum_helper_info);
 
-- 
cgit v1.2.3


From c4fa251f6f3ed00d59d0d8ee63bf346e6dd6b664 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:18:33 +0200
Subject: ALSA: ice1712: Use snd_ctl_enum_info()

... and reduce the open codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ews.c     | 16 ++--------------
 sound/pci/ice1712/ice1712.c | 27 ++++-----------------------
 2 files changed, 6 insertions(+), 37 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
index 817a1bc50a60..5cb587cf360e 100644
--- a/sound/pci/ice1712/ews.c
+++ b/sound/pci/ice1712/ews.c
@@ -580,13 +580,7 @@ static int snd_ice1712_ewx_io_sense_info(struct snd_kcontrol *kcontrol, struct s
 	static const char * const texts[2] = {
 		"+4dBu", "-10dBV",
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= 2)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_ice1712_ewx_io_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -903,13 +897,7 @@ static int snd_ice1712_6fire_select_input_info(struct snd_kcontrol *kcontrol, st
 	static const char * const texts[4] = {
 		"Internal", "Front Input", "Rear Input", "Wave Table"
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item >= 4)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
      
 static int snd_ice1712_6fire_select_input_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 206ed2cbcef9..48a0c330da24 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -1839,13 +1839,7 @@ static int snd_ice1712_pro_internal_clock_info(struct snd_kcontrol *kcontrol,
 		"96000",	/* 12: 7 */
 		"IEC958 Input",	/* 13: -- */
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 14;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 14, texts);
 }
 
 static int snd_ice1712_pro_internal_clock_get(struct snd_kcontrol *kcontrol,
@@ -1930,13 +1924,7 @@ static int snd_ice1712_pro_internal_clock_default_info(struct snd_kcontrol *kcon
 		"96000",	/* 12: 7 */
 		/* "IEC958 Input",	13: -- */
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 13;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 13, texts);
 }
 
 static int snd_ice1712_pro_internal_clock_default_get(struct snd_kcontrol *kcontrol,
@@ -2057,15 +2045,8 @@ static int snd_ice1712_pro_route_info(struct snd_kcontrol *kcontrol,
 		"IEC958 In L", "IEC958 In R", /* 9-10 */
 		"Digital Mixer", /* 11 - optional */
 	};
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items =
-		snd_ctl_get_ioffidx(kcontrol, &uinfo->id) < 2 ? 12 : 11;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	int num_items = snd_ctl_get_ioffidx(kcontrol, &uinfo->id) < 2 ? 12 : 11;
+	return snd_ctl_enum_info(uinfo, 1, num_items, texts);
 }
 
 static int snd_ice1712_pro_route_analog_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 597da2e4dfa04c8ee66b09fce931ab6825bc3e75 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:18:50 +0200
Subject: ALSA: ice1724: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/aureon.c       | 46 +++++++---------------------------------
 sound/pci/ice1712/ice1724.c      |  8 +------
 sound/pci/ice1712/maya44.c       | 20 ++---------------
 sound/pci/ice1712/phase.c        | 12 +----------
 sound/pci/ice1712/pontis.c       |  8 +------
 sound/pci/ice1712/prodigy192.c   | 18 ++--------------
 sound/pci/ice1712/prodigy_hifi.c | 11 ++--------
 sound/pci/ice1712/quartet.c      | 27 ++++-------------------
 sound/pci/ice1712/se.c           |  9 +-------
 9 files changed, 22 insertions(+), 137 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 3b3cf4ac9060..c9411dfff5a4 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -205,13 +205,7 @@ static int aureon_universe_inmux_info(struct snd_kcontrol *kcontrol,
 	static const char * const texts[3] =
 		{"Internal Aux", "Wavetable", "Rear Line-In"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int aureon_universe_inmux_get(struct snd_kcontrol *kcontrol,
@@ -1106,20 +1100,10 @@ static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in
 	};
 	struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 2;
-	if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) {
-		uinfo->value.enumerated.items = 8;
-		if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name, universe_texts[uinfo->value.enumerated.item]);
-	} else {
-		uinfo->value.enumerated.items = 5;
-		if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	}
-	return 0;
+	if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE)
+		return snd_ctl_enum_info(uinfo, 2, 8, universe_texts);
+	else
+		return snd_ctl_enum_info(uinfo, 2, 5, texts);
 }
 
 static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1167,16 +1151,10 @@ static int aureon_cs8415_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_
 		"CD",
 		"Coax"
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
 	if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71)
-		strcpy(uinfo->value.enumerated.name, prodigy_texts[uinfo->value.enumerated.item]);
+		return snd_ctl_enum_info(uinfo, 1, 2, prodigy_texts);
 	else
-		strcpy(uinfo->value.enumerated.name, aureon_texts[uinfo->value.enumerated.item]);
-	return 0;
+		return snd_ctl_enum_info(uinfo, 1, 2, aureon_texts);
 }
 
 static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1392,15 +1370,7 @@ static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_
 {
 	static const char * const texts[2] = { "128x", "64x"	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 08cb08ac85e6..f633e3bb4c43 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2049,13 +2049,7 @@ static int snd_vt1724_pro_route_info(struct snd_kcontrol *kcontrol,
 		"IEC958 In L", "IEC958 In R", /* 3-4 */
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 5;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 5, texts);
 }
 
 static inline int analog_route_shift(int idx)
diff --git a/sound/pci/ice1712/maya44.c b/sound/pci/ice1712/maya44.c
index 63aa39f06f02..7de25c4807fd 100644
--- a/sound/pci/ice1712/maya44.c
+++ b/sound/pci/ice1712/maya44.c
@@ -359,15 +359,7 @@ static int maya_rec_src_info(struct snd_kcontrol *kcontrol,
 {
 	static const char * const texts[] = { "Line", "Mic" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = ARRAY_SIZE(texts);
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
 }
 
 static int maya_rec_src_get(struct snd_kcontrol *kcontrol,
@@ -411,15 +403,7 @@ static int maya_pb_route_info(struct snd_kcontrol *kcontrol,
 		"Input 1", "Input 2", "Input 3", "Input 4"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = ARRAY_SIZE(texts);
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
 }
 
 static int maya_pb_route_shift(int idx)
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
index 0011e04f36a2..e9ca89c9174b 100644
--- a/sound/pci/ice1712/phase.c
+++ b/sound/pci/ice1712/phase.c
@@ -723,17 +723,7 @@ static int phase28_oversampling_info(struct snd_kcontrol *k,
 {
 	static const char * const texts[2] = { "128x", "64x"	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items -
-						1;
-	strcpy(uinfo->value.enumerated.name,
-		texts[uinfo->value.enumerated.item]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int phase28_oversampling_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index 5555eb4b2400..5101f40f6fbd 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -417,13 +417,7 @@ static int cs_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_inf
 		"Optical",	/* RXP1 */
 		"CD",		/* RXP2 */
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int cs_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index f3b491aa3e22..1eb151aaa965 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -284,15 +284,7 @@ static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol,
 {
 	static const char * const texts[2] = { "Line In", "Mic" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-
-        return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 
@@ -563,13 +555,7 @@ static int ak4114_input_sw_info(struct snd_kcontrol *kcontrol,
 {
 	static const char * const texts[2] = { "Toslink", "Coax" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-        return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 2261d1e49150..2697402b5195 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -537,7 +537,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol,
 static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_info *uinfo)
 {
-	static char* texts[32] = {
+	static const char * const texts[32] = {
 		"NULL", WM_AIN1, WM_AIN2, WM_AIN1 "+" WM_AIN2,
 		WM_AIN3, WM_AIN1 "+" WM_AIN3, WM_AIN2 "+" WM_AIN3,
 		WM_AIN1 "+" WM_AIN2 "+" WM_AIN3,
@@ -560,14 +560,7 @@ static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol,
 		WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 32;
-	if (uinfo->value.enumerated.item > 31)
-		uinfo->value.enumerated.item = 31;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 32, texts);
 }
 
 static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c
index 2c2df4b74e01..d4caf9d05922 100644
--- a/sound/pci/ice1712/quartet.c
+++ b/sound/pci/ice1712/quartet.c
@@ -46,7 +46,7 @@ struct qtet_kcontrol_private {
 	unsigned int bit;
 	void (*set_register)(struct snd_ice1712 *ice, unsigned int val);
 	unsigned int (*get_register)(struct snd_ice1712 *ice);
-	unsigned char * const texts[2];
+	const char * const texts[2];
 };
 
 enum {
@@ -554,17 +554,7 @@ static int qtet_ain12_enum_info(struct snd_kcontrol *kcontrol,
 {
 	static const char * const texts[3] =
 		{"Line In 1/2", "Mic", "Mic + Low-cut"};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = ARRAY_SIZE(texts);
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-			texts[uinfo->value.enumerated.item]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
 }
 
 static int qtet_ain12_sw_get(struct snd_kcontrol *kcontrol,
@@ -706,17 +696,8 @@ static int qtet_enum_info(struct snd_kcontrol *kcontrol,
 {
 	struct qtet_kcontrol_private private =
 		qtet_privates[kcontrol->private_value];
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = ARRAY_SIZE(private.texts);
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-			private.texts[uinfo->value.enumerated.item]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(private.texts),
+				 private.texts);
 }
 
 static int qtet_sw_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c
index ffd894bb4507..1c5d5b22c7a0 100644
--- a/sound/pci/ice1712/se.c
+++ b/sound/pci/ice1712/se.c
@@ -452,14 +452,7 @@ static int se200pci_cont_enum_info(struct snd_kcontrol *kc,
 	c = se200pci_get_enum_count(n);
 	if (!c)
 		return -EINVAL;
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = c;
-	if (uinfo->value.enumerated.item >= c)
-		uinfo->value.enumerated.item = c - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       se200pci_cont[n].member[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, c, se200pci_cont[n].member);
 }
 
 static int se200pci_cont_volume_get(struct snd_kcontrol *kc,
-- 
cgit v1.2.3


From f861237c80a07449abd351c04a6ba397418dc0ee Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:21:58 +0200
Subject: ALSA: korg1212: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/korg1212/korg1212.c | 26 +++++++-------------------
 1 file changed, 7 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 9fe549b2efdf..59d21c9401d2 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -444,9 +444,9 @@ static char *stateName[] = {
 	"Invalid"
 };
 
-static char *clockSourceTypeName[] = { "ADAT", "S/PDIF", "local" };
+static const char * const clockSourceTypeName[] = { "ADAT", "S/PDIF", "local" };
 
-static char *clockSourceName[] = {
+static const char * const clockSourceName[] = {
 	"ADAT at 44.1 kHz",
 	"ADAT at 48 kHz",
 	"S/PDIF at 44.1 kHz",
@@ -455,7 +455,7 @@ static char *clockSourceName[] = {
 	"local clock at 48 kHz"
 };
 
-static char *channelName[] = {
+static const char * const channelName[] = {
 	"ADAT-1",
 	"ADAT-2",
 	"ADAT-3",
@@ -1844,14 +1844,9 @@ static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol,
 static int snd_korg1212_control_route_info(struct snd_kcontrol *kcontrol,
 					   struct snd_ctl_elem_info *uinfo)
 {
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = (kcontrol->private_value >= 8) ? 2 : 1;
-	uinfo->value.enumerated.items = kAudioChannels;
-	if (uinfo->value.enumerated.item > kAudioChannels-1) {
-		uinfo->value.enumerated.item = kAudioChannels-1;
-	}
-	strcpy(uinfo->value.enumerated.name, channelName[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo,
+				 (kcontrol->private_value >= 8) ? 2 : 1,
+				 kAudioChannels, channelName);
 }
 
 static int snd_korg1212_control_route_get(struct snd_kcontrol *kcontrol,
@@ -1961,14 +1956,7 @@ static int snd_korg1212_control_put(struct snd_kcontrol *kcontrol,
 static int snd_korg1212_control_sync_info(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2) {
-		uinfo->value.enumerated.item = 2;
-	}
-	strcpy(uinfo->value.enumerated.name, clockSourceTypeName[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, clockSourceTypeName);
 }
 
 static int snd_korg1212_control_sync_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 08455ace3cafd9b0b5c35db3d89c4388f6d3a6fe Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:19:00 +0200
Subject: ALSA: pcxhr: Use snd_ctl_enum_info()

... and reduce the open codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr_mixer.c | 18 ++----------------
 1 file changed, 2 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index 95c9571780d8..63136c4f3f3d 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -660,14 +660,7 @@ static int pcxhr_audio_src_info(struct snd_kcontrol *kcontrol,
 		if (chip->mgr->board_has_mic)
 			i = 5;	/* Mic and MicroMix available */
 	}
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = i;
-	if (uinfo->value.enumerated.item > (i-1))
-		uinfo->value.enumerated.item = i-1;
-	strcpy(uinfo->value.enumerated.name,
-		texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, i, texts);
 }
 
 static int pcxhr_audio_src_get(struct snd_kcontrol *kcontrol,
@@ -756,14 +749,7 @@ static int pcxhr_clock_type_info(struct snd_kcontrol *kcontrol,
 		texts = textsPCXHR;
 		snd_BUG_ON(clock_items > (PCXHR_CLOCK_TYPE_MAX+1));
 	}
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = clock_items;
-	if (uinfo->value.enumerated.item >= clock_items)
-		uinfo->value.enumerated.item = clock_items-1;
-	strcpy(uinfo->value.enumerated.name,
-		texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, clock_items, texts);
 }
 
 static int pcxhr_clock_type_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 11c6ef7c8d439ef2bc3c95e5a4dcea449ab1f90f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:19:12 +0200
Subject: ALSA: rme32: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme32.c | 34 ++++++++++------------------------
 1 file changed, 10 insertions(+), 24 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 4afd3cab775b..6c60dcd2e5a1 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1608,30 +1608,24 @@ snd_rme32_info_inputtype_control(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
 	struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
-	static char *texts[4] = { "Optical", "Coaxial", "Internal", "XLR" };
+	static const char * const texts[4] = {
+		"Optical", "Coaxial", "Internal", "XLR"
+	};
+	int num_items;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
 	switch (rme32->pci->device) {
 	case PCI_DEVICE_ID_RME_DIGI32:
 	case PCI_DEVICE_ID_RME_DIGI32_8:
-		uinfo->value.enumerated.items = 3;
+		num_items = 3;
 		break;
 	case PCI_DEVICE_ID_RME_DIGI32_PRO:
-		uinfo->value.enumerated.items = 4;
+		num_items = 4;
 		break;
 	default:
 		snd_BUG();
-		break;
-	}
-	if (uinfo->value.enumerated.item >
-	    uinfo->value.enumerated.items - 1) {
-		uinfo->value.enumerated.item =
-		    uinfo->value.enumerated.items - 1;
+		return -EINVAL;
 	}
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, num_items, texts);
 }
 static int
 snd_rme32_get_inputtype_control(struct snd_kcontrol *kcontrol,
@@ -1695,20 +1689,12 @@ static int
 snd_rme32_info_clockmode_control(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = { "AutoSync", 
+	static const char * const texts[4] = { "AutoSync",
 				  "Internal 32.0kHz", 
 				  "Internal 44.1kHz", 
 				  "Internal 48.0kHz" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item > 3) {
-		uinfo->value.enumerated.item = 3;
-	}
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 static int
 snd_rme32_get_clockmode_control(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 9c30d46a0fb3b294faf1226025071d6e802a8c36 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:19:23 +0200
Subject: ALSA: rme96: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme96.c | 62 +++++++++++++++++++------------------------------------
 1 file changed, 21 insertions(+), 41 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 5a395c87c6fc..2f1a85185a16 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1884,39 +1884,38 @@ snd_rme96_put_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
 static int
 snd_rme96_info_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *_texts[5] = { "Optical", "Coaxial", "Internal", "XLR", "Analog" };
+	static const char * const _texts[5] = {
+		"Optical", "Coaxial", "Internal", "XLR", "Analog"
+	};
 	struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
-	char *texts[5] = { _texts[0], _texts[1], _texts[2], _texts[3], _texts[4] };
+	const char *texts[5] = {
+		_texts[0], _texts[1], _texts[2], _texts[3], _texts[4]
+	};
+	int num_items;
 	
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
 	switch (rme96->pci->device) {
 	case PCI_DEVICE_ID_RME_DIGI96:
 	case PCI_DEVICE_ID_RME_DIGI96_8:
-		uinfo->value.enumerated.items = 3;
+		num_items = 3;
 		break;
 	case PCI_DEVICE_ID_RME_DIGI96_8_PRO:
-		uinfo->value.enumerated.items = 4;
+		num_items = 4;
 		break;
 	case PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST:
 		if (rme96->rev > 4) {
 			/* PST */
-			uinfo->value.enumerated.items = 4;
+			num_items = 4;
 			texts[3] = _texts[4]; /* Analog instead of XLR */
 		} else {
 			/* PAD */
-			uinfo->value.enumerated.items = 5;
+			num_items = 5;
 		}
 		break;
 	default:
 		snd_BUG();
-		break;
-	}
-	if (uinfo->value.enumerated.item > uinfo->value.enumerated.items - 1) {
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+		return -EINVAL;
 	}
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, num_items, texts);
 }
 static int
 snd_rme96_get_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2002,16 +2001,9 @@ snd_rme96_put_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 static int
 snd_rme96_info_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = { "AutoSync", "Internal", "Word" };
+	static const char * const texts[3] = { "AutoSync", "Internal", "Word" };
 	
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2) {
-		uinfo->value.enumerated.item = 2;
-	}
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 static int
 snd_rme96_get_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2041,16 +2033,11 @@ snd_rme96_put_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 static int
 snd_rme96_info_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = { "0 dB", "-6 dB", "-12 dB", "-18 dB" };
+	static const char * const texts[4] = {
+		"0 dB", "-6 dB", "-12 dB", "-18 dB"
+	};
 	
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item > 3) {
-		uinfo->value.enumerated.item = 3;
-	}
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 static int
 snd_rme96_get_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2081,16 +2068,9 @@ snd_rme96_put_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_
 static int
 snd_rme96_info_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = { "1+2", "3+4", "5+6", "7+8" };
+	static const char * const texts[4] = { "1+2", "3+4", "5+6", "7+8" };
 	
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item > 3) {
-		uinfo->value.enumerated.item = 3;
-	}
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 static int
 snd_rme96_get_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-- 
cgit v1.2.3


From 8d678da9f0afbb951778369510c09b99de608c24 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:19:34 +0200
Subject: ALSA: hdsp: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 175 +++++++++++++++--------------------------------
 1 file changed, 57 insertions(+), 118 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 7646ba1664eb..2eb8baf7b828 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -1680,16 +1680,13 @@ static int hdsp_set_spdif_input(struct hdsp *hdsp, int in)
 
 static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = {"Optical", "Coaxial", "Internal", "AES"};
+	static const char * const texts[4] = {
+		"Optical", "Coaxial", "Internal", "AES"
+	};
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = ((hdsp->io_type == H9632) ? 4 : 3);
-	if (uinfo->value.enumerated.item > ((hdsp->io_type == H9632) ? 3 : 2))
-		uinfo->value.enumerated.item = ((hdsp->io_type == H9632) ? 3 : 2);
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 4 : 3,
+				 texts);
 }
 
 static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1786,16 +1783,14 @@ static int snd_hdsp_put_toggle_setting(struct snd_kcontrol *kcontrol,
 
 static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};
+	static const char * const texts[] = {
+		"32000", "44100", "48000", "64000", "88200", "96000",
+		"None", "128000", "176400", "192000"
+	};
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 10 : 7,
+				 texts);
 }
 
 static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1872,14 +1867,13 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	static const char * const texts[] = {
+		"32000", "44100", "48000", "64000", "88200", "96000",
+		"None", "128000", "176400", "192000"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 10 : 7,
+				 texts);
 }
 
 static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1940,15 +1934,9 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp)
 
 static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"Master", "Slave" };
+	static const char * const texts[] = {"Master", "Slave" };
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2049,19 +2037,16 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode)
 
 static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" };
+	static const char * const texts[] = {
+		"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+		"Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+		"Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz",
+		"Internal 192.0 KHz"
+	};
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	if (hdsp->io_type == H9632)
-	    uinfo->value.enumerated.items = 10;
-	else
-	    uinfo->value.enumerated.items = 7;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 10 : 7,
+				 texts);
 }
 
 static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2165,15 +2150,9 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode)
 
 static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"};
+	static const char * const texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2250,15 +2229,9 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode)
 
 static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"};
+	static const char * const texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2335,15 +2308,9 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode)
 
 static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"0 dB", "-6 dB", "-12 dB"};
+	static const char * const texts[] = {"0 dB", "-6 dB", "-12 dB"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2439,31 +2406,28 @@ static int hdsp_set_pref_sync_ref(struct hdsp *hdsp, int pref)
 
 static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" };
+	static const char * const texts[] = {
+		"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3"
+	};
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
+	int num_items;
 
 	switch (hdsp->io_type) {
 	case Digiface:
 	case H9652:
-		uinfo->value.enumerated.items = 6;
+		num_items = 6;
 		break;
 	case Multiface:
-		uinfo->value.enumerated.items = 4;
+		num_items = 4;
 		break;
 	case H9632:
-		uinfo->value.enumerated.items = 3;
+		num_items = 3;
 		break;
 	default:
 		return -EINVAL;
 	}
 
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, num_items, texts);
 }
 
 static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2543,15 +2507,11 @@ static int hdsp_autosync_ref(struct hdsp *hdsp)
 
 static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" };
+	static const char * const texts[] = {
+		"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3"
+	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 7;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 7, texts);
 }
 
 static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -2738,14 +2698,9 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 
 static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No Lock", "Lock", "Sync" };
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	static const char * const texts[] = {"No Lock", "Lock", "Sync" };
+
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int hdsp_wc_sync_check(struct hdsp *hdsp)
@@ -3101,15 +3056,11 @@ static int snd_hdsp_put_rpm_input12(struct snd_kcontrol *kcontrol, struct snd_ct
 
 static int snd_hdsp_info_rpm_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"Phono +6dB", "Phono 0dB", "Phono -6dB", "Line 0dB", "Line -6dB"};
+	static const char * const texts[] = {
+		"Phono +6dB", "Phono 0dB", "Phono -6dB", "Line 0dB", "Line -6dB"
+	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 5;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 5, texts);
 }
 
 
@@ -3234,15 +3185,9 @@ static int snd_hdsp_put_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl
 
 static int snd_hdsp_info_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"On", "Off"};
+	static const char * const texts[] = {"On", "Off"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 
@@ -3291,15 +3236,9 @@ static int snd_hdsp_put_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd
 
 static int snd_hdsp_info_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"On", "Off"};
+	static const char * const texts[] = {"On", "Off"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static struct snd_kcontrol_new snd_hdsp_rpm_controls[] = {
-- 
cgit v1.2.3


From c69a637b4df37fc5a011a89e422636ea393af5b0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:19:41 +0200
Subject: ALSA: hdspm: Use snd_ctl_enum_info()

... and reduce the open codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdspm.c | 13 +------------
 1 file changed, 1 insertion(+), 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 52d86af3ef2d..7f7277bfb66a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -2645,18 +2645,7 @@ static int hdspm_set_clock_source(struct hdspm * hdspm, int mode)
 static int snd_hdspm_info_clock_source(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 9;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-		    uinfo->value.enumerated.items - 1;
-
-	strcpy(uinfo->value.enumerated.name,
-	       texts_freq[uinfo->value.enumerated.item+1]);
-
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 9, texts_freq + 1);
 }
 
 static int snd_hdspm_get_clock_source(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 7298ece7a26753b073a9ce5f979a4942d3904d44 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:19:53 +0200
Subject: ALSA: rme9652: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to text arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/rme9652.c | 58 ++++++++++++++-------------------------------
 1 file changed, 18 insertions(+), 40 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index fa9a2a8dce5a..6521521853b8 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -920,15 +920,9 @@ static int rme9652_set_adat1_input(struct snd_rme9652 *rme9652, int internal)
 
 static int snd_rme9652_info_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2] = {"ADAT1", "Internal"};
+	static const char * const texts[2] = {"ADAT1", "Internal"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_rme9652_get_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -991,15 +985,9 @@ static int rme9652_set_spdif_input(struct snd_rme9652 *rme9652, int in)
 
 static int snd_rme9652_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = {"ADAT1", "Coaxial", "Internal"};
+	static const char * const texts[3] = {"ADAT1", "Coaxial", "Internal"};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_rme9652_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1140,15 +1128,11 @@ static int rme9652_set_sync_mode(struct snd_rme9652 *rme9652, int mode)
 
 static int snd_rme9652_info_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = {"AutoSync", "Master", "Word Clock"};
+	static const char * const texts[3] = {
+		"AutoSync", "Master", "Word Clock"
+	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 3;
-	if (uinfo->value.enumerated.item > 2)
-		uinfo->value.enumerated.item = 2;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 3, texts);
 }
 
 static int snd_rme9652_get_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1231,16 +1215,14 @@ static int rme9652_set_sync_pref(struct snd_rme9652 *rme9652, int pref)
 
 static int snd_rme9652_info_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = {"IEC958 In", "ADAT1 In", "ADAT2 In", "ADAT3 In"};
+	static const char * const texts[4] = {
+		"IEC958 In", "ADAT1 In", "ADAT2 In", "ADAT3 In"
+	};
 	struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1,
+				 rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3,
+				 texts);
 }
 
 static int snd_rme9652_get_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1392,15 +1374,11 @@ static int snd_rme9652_get_spdif_rate(struct snd_kcontrol *kcontrol, struct snd_
 
 static int snd_rme9652_info_adat_sync(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[4] = {"No Lock", "Lock", "No Lock Sync", "Lock Sync"};
+	static const char * const texts[4] = {
+		"No Lock", "Lock", "No Lock Sync", "Lock Sync"
+	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
 }
 
 static int snd_rme9652_get_adat_sync(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-- 
cgit v1.2.3


From 3e4bc5b78e5516585941c7888287ed50a5090bf4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:20:07 +0200
Subject: ALSA: sonicvibes: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to the text array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/sonicvibes.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 5b0d317cc9a6..313a7328bf9c 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -918,17 +918,11 @@ static int snd_sonicvibes_pcm(struct sonicvibes *sonic, int device,
 
 static int snd_sonicvibes_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[7] = {
+	static const char * const texts[7] = {
 		"CD", "PCM", "Aux1", "Line", "Aux0", "Mic", "Mix"
 	};
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 2;
-	uinfo->value.enumerated.items = 7;
-	if (uinfo->value.enumerated.item >= 7)
-		uinfo->value.enumerated.item = 6;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 2, 7, texts);
 }
 
 static int snd_sonicvibes_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-- 
cgit v1.2.3


From 9883ab91e3ba5229bfe2d6e7f6ff497a2d03d4d2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Oct 2014 18:20:18 +0200
Subject: ALSA: via82xx: Use snd_ctl_enum_info()

... and reduce the open codes.  Also add missing const to the text array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/via82xx.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index ecedf4dbfa2a..e088467fb736 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1610,16 +1610,10 @@ static int snd_via8233_capture_source_info(struct snd_kcontrol *kcontrol,
 	/* formerly they were "Line" and "Mic", but it looks like that they
 	 * have nothing to do with the actual physical connections...
 	 */
-	static char *texts[2] = {
+	static const char * const texts[2] = {
 		"Input1", "Input2"
 	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= 2)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
 }
 
 static int snd_via8233_capture_source_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From cf6814f2b5014ed5bbdef764a42e4abaa09b3a2e Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Tue, 21 Oct 2014 16:28:47 +0530
Subject: ALSA: ctxfi: remove unused variable

As of now the pointer to struct dai is not being used anywhere in the
function. So it is safe to remove the variable.
If we are ever doing anything with the container_of(daio, struct dai, daio),
then at that time we can again add the variable.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctatc.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 454659074390..632e843fa95e 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1145,7 +1145,6 @@ static int atc_release_resources(struct ct_atc *atc)
 	int i;
 	struct daio_mgr *daio_mgr = NULL;
 	struct dao *dao = NULL;
-	struct dai *dai = NULL;
 	struct daio *daio = NULL;
 	struct sum_mgr *sum_mgr = NULL;
 	struct src_mgr *src_mgr = NULL;
@@ -1172,9 +1171,6 @@ static int atc_release_resources(struct ct_atc *atc)
 				dao = container_of(daio, struct dao, daio);
 				dao->ops->clear_left_input(dao);
 				dao->ops->clear_right_input(dao);
-			} else {
-				dai = container_of(daio, struct dai, daio);
-				/* some thing to do for dai ... */
 			}
 			daio_mgr->put_daio(daio_mgr, daio);
 		}
-- 
cgit v1.2.3


From 66797f36fd17e8975f4a3449aed895cda952c0ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Oct 2014 15:15:44 +0100
Subject: ALSA: hda - Pass printf argument directly to request_module()

request_module() handles the printf style arguments, so we don't have
to render strings in the caller side.  Not only it reduces the
unnecessary temporary string buffer, it's even safer from the security
POV.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 259fbeaa37bd..0025bf4c2f44 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -966,14 +966,12 @@ find_codec_preset(struct hda_codec *codec)
 	mutex_unlock(&preset_mutex);
 
 	if (mod_requested < HDA_MODREQ_MAX_COUNT) {
-		char name[32];
 		if (!mod_requested)
-			snprintf(name, sizeof(name), "snd-hda-codec-id:%08x",
-				 codec->vendor_id);
+			request_module("snd-hda-codec-id:%08x",
+				       codec->vendor_id);
 		else
-			snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*",
-				 (codec->vendor_id >> 16) & 0xffff);
-		request_module(name);
+			request_module("snd-hda-codec-id:%04x*",
+				       (codec->vendor_id >> 16) & 0xffff);
 		mod_requested++;
 		goto again;
 	}
-- 
cgit v1.2.3


From 326f0480b7f8504c4f594c4f36ab7874e17780bc Mon Sep 17 00:00:00 2001
From: Aya Mahfouz <mahfouz.saif.elyazal@gmail.com>
Date: Tue, 28 Oct 2014 14:27:44 +0200
Subject: ALSA: pcxhr: convert timeval to ktime_t

This patch is concerned with migrating the time variables in the pcxhr
module found in the sound driver. The changes are concerend with the
y2038 problem where timeval will overflow in the year 2038. ktime_t
was used instead of timeval to get the wall time. The difference
is displayed now in nanoseconds instead of microseconds.

Signed-off-by: Aya Mahfouz <mahfouz.saif.elyazal@gmail.com>
Reviewed-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.c      | 10 ++++++----
 sound/pci/pcxhr/pcxhr_core.c | 10 ++++++----
 2 files changed, 12 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index b854fc5e01f5..7c33c973dbd5 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -711,8 +711,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr)
 	int playback_mask = 0;
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
-	struct timeval my_tv1, my_tv2;
-	do_gettimeofday(&my_tv1);
+	ktime_t start_time, stop_time, diff_time;
+
+	start_time = ktime_get();
 #endif
 	mutex_lock(&mgr->setup_mutex);
 
@@ -823,9 +824,10 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr)
 	mutex_unlock(&mgr->setup_mutex);
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
-	do_gettimeofday(&my_tv2);
+	stop_time = ktime_get();
+	diff_time = ktime_sub(stop_time, start_time);
 	dev_dbg(&mgr->pci->dev, "***TRIGGER START*** TIME = %ld (err = %x)\n",
-		    (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
+		    (long)(ktime_to_ns(diff_time)), err);
 #endif
 }
 
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index a584acb61c00..181f7729d409 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -910,8 +910,9 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask,
 	int audio_mask;
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
-	struct timeval my_tv1, my_tv2;
-	do_gettimeofday(&my_tv1);
+	ktime_t start_time, stop_time, diff_time;
+
+	start_time = ktime_get();
 #endif
 	audio_mask = (playback_mask |
 		      (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET));
@@ -960,9 +961,10 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask,
 			return err;
 	}
 #ifdef CONFIG_SND_DEBUG_VERBOSE
-	do_gettimeofday(&my_tv2);
+	stop_time = ktime_get();
+	diff_time = ktime_sub(stop_time, start_time);
 	dev_dbg(&mgr->pci->dev, "***SET PIPE STATE*** TIME = %ld (err = %x)\n",
-		    (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
+			(long)(ktime_to_ns(diff_time)), err);
 #endif
 	return 0;
 }
-- 
cgit v1.2.3


From 2e6705c09065ecb357140e44d12dc32274b1a723 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Oct 2014 17:16:51 +0100
Subject: ALSA: ctxfi: Kill the rest snd_print*()

Use the standard dev_*() instead.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctatc.c   | 6 ++++--
 sound/pci/ctxfi/cttimer.c | 4 ++--
 2 files changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 632e843fa95e..977a59855fa6 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -438,7 +438,9 @@ atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm)
 	position = src->ops->get_ca(src);
 
 	if (position < apcm->vm_block->addr) {
-		snd_printdd("ctxfi: bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n", position, apcm->vm_block->addr, apcm->vm_block->size);
+		dev_dbg(atc->card->dev,
+			"bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n",
+			position, apcm->vm_block->addr, apcm->vm_block->size);
 		position = apcm->vm_block->addr;
 	}
 
@@ -1295,7 +1297,7 @@ static int atc_identify_card(struct ct_atc *atc, unsigned int ssid)
 			atc->model = CT20K2_UNKNOWN;
 	}
 	atc->model_name = ct_subsys_name[atc->model];
-	snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n",
+	dev_info(atc->card->dev, "chip %s model %s (%04x:%04x) is found\n",
 		   atc->chip_name, atc->model_name,
 		   vendor_id, device_id);
 	return 0;
diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c
index 03fb909085af..a5d460453d7b 100644
--- a/sound/pci/ctxfi/cttimer.c
+++ b/sound/pci/ctxfi/cttimer.c
@@ -421,12 +421,12 @@ struct ct_timer *ct_timer_new(struct ct_atc *atc)
 	atimer->atc = atc;
 	hw = atc->hw;
 	if (!use_system_timer && hw->set_timer_irq) {
-		snd_printd(KERN_INFO "ctxfi: Use xfi-native timer\n");
+		dev_info(atc->card->dev, "Use xfi-native timer\n");
 		atimer->ops = &ct_xfitimer_ops;
 		hw->irq_callback_data = atimer;
 		hw->irq_callback = ct_timer_interrupt;
 	} else {
-		snd_printd(KERN_INFO "ctxfi: Use system timer\n");
+		dev_info(atc->card->dev, "Use system timer\n");
 		atimer->ops = &ct_systimer_ops;
 	}
 	return atimer;
-- 
cgit v1.2.3


From f48a6df28239f5bf35d80e43e580261d2298395a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Oct 2014 17:22:45 +0100
Subject: ALSA: pcxhr: Kill the rest snd_print*()

Use the standard dev_*() instead.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.c | 42 +++++++++++++++++++++---------------------
 1 file changed, 21 insertions(+), 21 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 7c33c973dbd5..a60293015267 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -501,10 +501,10 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr,
 /*
  *  start or stop playback/capture substream
  */
-static int pcxhr_set_stream_state(struct pcxhr_stream *stream)
+static int pcxhr_set_stream_state(struct snd_pcxhr *chip,
+				  struct pcxhr_stream *stream)
 {
 	int err;
-	struct snd_pcxhr *chip;
 	struct pcxhr_rmh rmh;
 	int stream_mask, start;
 
@@ -512,8 +512,8 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream)
 		start = 1;
 	else {
 		if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) {
-			snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state "
-				   "CANNOT be stopped\n");
+			dev_err(chip->card->dev,
+				"pcxhr_set_stream_state CANNOT be stopped\n");
 			return -EINVAL;
 		}
 		start = 0;
@@ -560,6 +560,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream)
 	struct pcxhr_rmh rmh;
 	unsigned int header;
 
+	chip = snd_pcm_substream_chip(stream->substream);
 	switch (stream->format) {
 	case SNDRV_PCM_FORMAT_U8:
 		header = HEADER_FMT_BASE_LIN;
@@ -582,11 +583,10 @@ static int pcxhr_set_format(struct pcxhr_stream *stream)
 		header = HEADER_FMT_BASE_FLOAT | HEADER_FMT_INTEL;
 		break;
 	default:
-		snd_printk(KERN_ERR
-			   "error pcxhr_set_format() : unknown format\n");
+		dev_err(chip->card->dev,
+			"error pcxhr_set_format() : unknown format\n");
 		return -EINVAL;
 	}
-	chip = snd_pcm_substream_chip(stream->substream);
 
 	sample_rate = chip->mgr->sample_rate;
 	if (sample_rate <= 32000 && sample_rate !=0) {
@@ -643,11 +643,11 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream)
 	is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE);
 	stream_num = is_capture ? 0 : subs->number;
 
-	snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : "
-		    "addr(%p) bytes(%zx) subs(%d)\n",
-		    is_capture ? 'c' : 'p',
-		    chip->chip_idx, (void *)(long)subs->runtime->dma_addr,
-		    subs->runtime->dma_bytes, subs->number);
+	dev_dbg(chip->card->dev,
+		"pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n",
+		is_capture ? 'c' : 'p',
+		chip->chip_idx, (void *)(long)subs->runtime->dma_addr,
+		subs->runtime->dma_bytes, subs->number);
 
 	pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS);
 	pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio,
@@ -687,7 +687,7 @@ static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream,
 		*sample_count = ((snd_pcm_uframes_t)rmh.stat[0]) << 24;
 		*sample_count += (snd_pcm_uframes_t)rmh.stat[1];
 	}
-	snd_printdd("PIPE_SAMPLE_COUNT = %lx\n", *sample_count);
+	dev_dbg(chip->card->dev, "PIPE_SAMPLE_COUNT = %lx\n", *sample_count);
 	return err;
 }
 #endif
@@ -779,12 +779,12 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr)
 		for (j = 0; j < chip->nb_streams_capt; j++) {
 			stream = &chip->capture_stream[j];
 			if (pcxhr_stream_scheduled_get_pipe(stream, &pipe))
-				err = pcxhr_set_stream_state(stream);
+				err = pcxhr_set_stream_state(chip, stream);
 		}
 		for (j = 0; j < chip->nb_streams_play; j++) {
 			stream = &chip->playback_stream[j];
 			if (pcxhr_stream_scheduled_get_pipe(stream, &pipe))
-				err = pcxhr_set_stream_state(stream);
+				err = pcxhr_set_stream_state(chip, stream);
 		}
 	}
 
@@ -839,12 +839,12 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
 {
 	struct pcxhr_stream *stream;
 	struct snd_pcm_substream *s;
+	struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		snd_printdd("SNDRV_PCM_TRIGGER_START\n");
+		dev_dbg(chip->card->dev, "SNDRV_PCM_TRIGGER_START\n");
 		if (snd_pcm_stream_linked(subs)) {
-			struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
 			snd_pcm_group_for_each_entry(s, subs) {
 				if (snd_pcm_substream_chip(s) != chip)
 					continue;
@@ -856,7 +856,7 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
 			pcxhr_start_linked_stream(chip->mgr);
 		} else {
 			stream = subs->runtime->private_data;
-			snd_printdd("Only one Substream %c %d\n",
+			dev_dbg(chip->card->dev, "Only one Substream %c %d\n",
 				    stream->pipe->is_capture ? 'C' : 'P',
 				    stream->pipe->first_audio);
 			if (pcxhr_set_format(stream))
@@ -865,17 +865,17 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
 				return -EINVAL;
 
 			stream->status = PCXHR_STREAM_STATUS_SCHEDULE_RUN;
-			if (pcxhr_set_stream_state(stream))
+			if (pcxhr_set_stream_state(chip, stream))
 				return -EINVAL;
 			stream->status = PCXHR_STREAM_STATUS_RUNNING;
 		}
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
-		snd_printdd("SNDRV_PCM_TRIGGER_STOP\n");
+		dev_dbg(chip->card->dev, "SNDRV_PCM_TRIGGER_STOP\n");
 		snd_pcm_group_for_each_entry(s, subs) {
 			stream = s->runtime->private_data;
 			stream->status = PCXHR_STREAM_STATUS_SCHEDULE_STOP;
-			if (pcxhr_set_stream_state(stream))
+			if (pcxhr_set_stream_state(chip, stream))
 				return -EINVAL;
 			snd_pcm_trigger_done(s, subs);
 		}
-- 
cgit v1.2.3


From f994cb3a09a5f2018c286f854c10277132f4a9a5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Oct 2014 17:36:50 +0100
Subject: ALSA: au88x0: Kill the rest snd_print*()

Use the standard dev_*() instead.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index e9c3833f6d44..996369134ea8 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -314,7 +314,7 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_VORTEX_SYNTH,
 			       sizeof(snd_vortex_synth_arg_t), &wave) < 0
 	    || wave == NULL) {
-		snd_printk(KERN_ERR "Can't initialize Aureal wavetable synth\n");
+		dev_err(card->dev, "Can't initialize Aureal wavetable synth\n");
 	} else {
 		snd_vortex_synth_arg_t *arg;
 
-- 
cgit v1.2.3


From a11e9b168646cfc5d3b8d605d430d7e4ff267d72 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Oct 2014 15:06:01 +0100
Subject: ALSA: hda - Correct kerneldoc comments

Complete the missing parameters and fix anything wrong there.
Just comment changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c |  15 +++++-
 sound/pci/hda/hda_codec.c       | 114 ++++++++++++++++++++++++++++++++++++++--
 sound/pci/hda/hda_eld.c         |   2 +-
 sound/pci/hda/hda_jack.c        |  21 ++++++++
 4 files changed, 146 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index fcc5e478c9a1..7388958b01af 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -464,8 +464,12 @@ EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr);
 
 /**
  * hda_get_input_pin_label - Give a label for the given input pin
+ * @codec: the HDA codec
+ * @item: ping config item to refer
+ * @pin: the pin NID
+ * @check_location: flag to add the jack location prefix
  *
- * When check_location is true, the function checks the pin location
+ * When @check_location is true, the function checks the pin location
  * for mic and line-in pins, and set an appropriate prefix like "Front",
  * "Rear", "Internal".
  */
@@ -550,6 +554,9 @@ static int check_mic_location_need(struct hda_codec *codec,
 
 /**
  * hda_get_autocfg_input_label - Get a label for the given input
+ * @codec: the HDA codec
+ * @cfg: the parsed pin configuration
+ * @input: the input index number
  *
  * Get a label for the given input pin defined by the autocfg item.
  * Unlike hda_get_input_pin_label(), this function checks all inputs
@@ -677,6 +684,12 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
 
 /**
  * snd_hda_get_pin_label - Get a label for the given I/O pin
+ * @codec: the HDA codec
+ * @nid: pin NID
+ * @cfg: the parsed pin configuration
+ * @label: the string buffer to store
+ * @maxlen: the max length of string buffer (including termination)
+ * @indexp: the pointer to return the index number (for multiple ctls)
  *
  * Get a label for the given pin.  This function works for both input and
  * output pins.  When @cfg is given as non-NULL, the function tries to get
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0025bf4c2f44..e15254204c72 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -416,7 +416,6 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid)
  * snd_hda_get_conn_list - get connection list
  * @codec: the HDA codec
  * @nid: NID to parse
- * @len: number of connection list entries
  * @listp: the pointer to store NID list
  *
  * Parses the connection list of the given widget and stores the pointer
@@ -2004,6 +2003,7 @@ EXPORT_SYMBOL_GPL(query_amp_caps);
  * @codec: the HD-audio codec
  * @nid: the NID to query
  * @dir: either #HDA_INPUT or #HDA_OUTPUT
+ * @bits: bit mask to check the result
  *
  * Check whether the widget has the given amp capability for the direction.
  */
@@ -2023,7 +2023,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_caps);
  * snd_hda_override_amp_caps - Override the AMP capabilities
  * @codec: the CODEC to clean up
  * @nid: the NID to clean up
- * @direction: either #HDA_INPUT or #HDA_OUTPUT
+ * @dir: either #HDA_INPUT or #HDA_OUTPUT
  * @caps: the capability bits to set
  *
  * Override the cached AMP caps bits value by the given one.
@@ -2320,6 +2320,8 @@ static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir,
 
 /**
  * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer
+ * @kcontrol: referred ctl element
+ * @uinfo: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -2381,6 +2383,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid,
 
 /**
  * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -2406,6 +2410,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_get);
 
 /**
  * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -2436,6 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put);
 
 /**
  * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume
+ * @kcontrol: ctl element
+ * @op_flag: operation flag
+ * @size: byte size of input TLV
+ * @_tlv: TLV data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -3012,6 +3022,8 @@ EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook);
 
 /**
  * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch
+ * @kcontrol: referred ctl element
+ * @uinfo: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -3031,6 +3043,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_info);
 
 /**
  * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -3057,6 +3071,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get);
 
 /**
  * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
@@ -3100,6 +3116,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put);
 
 /**
  * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_BIND_MUTE*() macros.
@@ -3123,6 +3141,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_get);
 
 /**
  * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_BIND_MUTE*() macros.
@@ -3153,6 +3173,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_put);
 
 /**
  * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control
+ * @kcontrol: referred ctl element
+ * @uinfo: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros.
@@ -3176,6 +3198,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_info);
 
 /**
  * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros.
@@ -3199,6 +3223,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_get);
 
 /**
  * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros.
@@ -3228,6 +3254,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_put);
 
 /**
  * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control
+ * @kcontrol: ctl element
+ * @op_flag: operation flag
+ * @size: byte size of input TLV
+ * @tlv: TLV data
  *
  * The control element is supposed to have the private_value field
  * set up via HDA_BIND_VOL() macro.
@@ -4305,6 +4335,7 @@ static struct hda_rate_tbl rate_bits[] = {
  * @channels: the number of channels
  * @format: the PCM format (SNDRV_PCM_FORMAT_XXX)
  * @maxbps: the max. bps
+ * @spdif_ctls: HD-audio SPDIF status bits (0 if irrelevant)
  *
  * Calculate the format bitset from the given rate, channels and th PCM format.
  *
@@ -4980,6 +5011,7 @@ static void __snd_hda_power_down(struct hda_codec *codec)
  * snd_hda_power_save - Power-up/down/sync the codec
  * @codec: HD-audio codec
  * @delta: the counter delta to change
+ * @d3wait: sync for D3 transition complete
  *
  * Change the power-up counter via @delta, and power up or down the hardware
  * appropriately.  For the power-down, queue to the delayed action.
@@ -5055,6 +5087,10 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power);
 
 /**
  * snd_hda_ch_mode_info - Info callback helper for the channel mode enum
+ * @codec: the HDA codec
+ * @uinfo: pointer to get/store the data
+ * @chmode: channel mode array
+ * @num_chmodes: channel mode array size
  */
 int snd_hda_ch_mode_info(struct hda_codec *codec,
 			 struct snd_ctl_elem_info *uinfo,
@@ -5074,6 +5110,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info);
 
 /**
  * snd_hda_ch_mode_get - Get callback helper for the channel mode enum
+ * @codec: the HDA codec
+ * @ucontrol: pointer to get/store the data
+ * @chmode: channel mode array
+ * @num_chmodes: channel mode array size
+ * @max_channels: max number of channels
  */
 int snd_hda_ch_mode_get(struct hda_codec *codec,
 			struct snd_ctl_elem_value *ucontrol,
@@ -5095,6 +5136,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get);
 
 /**
  * snd_hda_ch_mode_put - Put callback helper for the channel mode enum
+ * @codec: the HDA codec
+ * @ucontrol: pointer to get/store the data
+ * @chmode: channel mode array
+ * @num_chmodes: channel mode array size
+ * @max_channelsp: pointer to store the max channels
  */
 int snd_hda_ch_mode_put(struct hda_codec *codec,
 			struct snd_ctl_elem_value *ucontrol,
@@ -5123,6 +5169,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put);
 
 /**
  * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum
+ * @imux: imux helper object
+ * @uinfo: pointer to get/store the data
  */
 int snd_hda_input_mux_info(const struct hda_input_mux *imux,
 			   struct snd_ctl_elem_info *uinfo)
@@ -5144,6 +5192,11 @@ EXPORT_SYMBOL_GPL(snd_hda_input_mux_info);
 
 /**
  * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum
+ * @codec: the HDA codec
+ * @imux: imux helper object
+ * @ucontrol: pointer to get/store the data
+ * @nid: input mux NID
+ * @cur_val: pointer to get/store the current imux value
  */
 int snd_hda_input_mux_put(struct hda_codec *codec,
 			  const struct hda_input_mux *imux,
@@ -5168,7 +5221,13 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 EXPORT_SYMBOL_GPL(snd_hda_input_mux_put);
 
 
-/*
+/**
+ * snd_hda_enum_helper_info - Helper for simple enum ctls
+ * @kcontrol: ctl element
+ * @uinfo: pointer to get/store the data
+ * @num_items: number of enum items
+ * @texts: enum item string array
+ *
  * process kcontrol info callback of a simple string enum array
  * when @num_items is 0 or @texts is NULL, assume a boolean enum array
  */
@@ -5257,6 +5316,8 @@ EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify);
 
 /**
  * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
  */
 int snd_hda_multi_out_dig_open(struct hda_codec *codec,
 			       struct hda_multi_out *mout)
@@ -5273,6 +5334,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_open);
 
 /**
  * snd_hda_multi_out_dig_prepare - prepare the digital out stream
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
+ * @stream_tag: stream tag to assign
+ * @format: format id to assign
+ * @substream: PCM substream to assign
  */
 int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 				  struct hda_multi_out *mout,
@@ -5289,6 +5355,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_prepare);
 
 /**
  * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
  */
 int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
 				  struct hda_multi_out *mout)
@@ -5302,6 +5370,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_cleanup);
 
 /**
  * snd_hda_multi_out_dig_close - release the digital out stream
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
  */
 int snd_hda_multi_out_dig_close(struct hda_codec *codec,
 				struct hda_multi_out *mout)
@@ -5315,6 +5385,10 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close);
 
 /**
  * snd_hda_multi_out_analog_open - open analog outputs
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
+ * @substream: PCM substream to assign
+ * @hinfo: PCM information to assign
  *
  * Open analog outputs and set up the hw-constraints.
  * If the digital outputs can be opened as slave, open the digital
@@ -5365,6 +5439,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open);
 
 /**
  * snd_hda_multi_out_analog_prepare - Preapre the analog outputs.
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
+ * @stream_tag: stream tag to assign
+ * @format: format id to assign
+ * @substream: PCM substream to assign
  *
  * Set up the i/o for analog out.
  * When the digital out is available, copy the front out to digital out, too.
@@ -5442,6 +5521,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_prepare);
 
 /**
  * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out
+ * @codec: the HDA codec
+ * @mout: hda_multi_out object
  */
 int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 				     struct hda_multi_out *mout)
@@ -5473,6 +5554,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup);
 
 /**
  * snd_hda_get_default_vref - Get the default (mic) VREF pin bits
+ * @codec: the HDA codec
+ * @pin: referred pin NID
  *
  * Guess the suitable VREF pin bits to be set as the pin-control value.
  * Note: the function doesn't set the AC_PINCTL_IN_EN bit.
@@ -5498,7 +5581,12 @@ unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin)
 }
 EXPORT_SYMBOL_GPL(snd_hda_get_default_vref);
 
-/* correct the pin ctl value for matching with the pin cap */
+/**
+ * snd_hda_correct_pin_ctl - correct the pin ctl value for matching with the pin cap
+ * @codec: the HDA codec
+ * @pin: referred pin NID
+ * @val: pin ctl value to audit
+ */
 unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec,
 				     hda_nid_t pin, unsigned int val)
 {
@@ -5549,6 +5637,19 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl);
 
+/**
+ * _snd_hda_pin_ctl - Helper to set pin ctl value
+ * @codec: the HDA codec
+ * @pin: referred pin NID
+ * @val: pin control value to set
+ * @cached: access over codec pinctl cache or direct write
+ *
+ * This function is a helper to set a pin ctl value more safely.
+ * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the
+ * value in pin target array via snd_hda_codec_set_pin_target(), then
+ * actually writes the value via either snd_hda_codec_update_cache() or
+ * snd_hda_codec_write() depending on @cached flag.
+ */
 int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int val, bool cached)
 {
@@ -5565,6 +5666,11 @@ EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl);
 
 /**
  * snd_hda_add_imux_item - Add an item to input_mux
+ * @codec: the HDA codec
+ * @imux: imux helper object
+ * @label: the name of imux item to assign
+ * @index: index number of imux item to assign
+ * @type_idx: pointer to store the resultant label index
  *
  * When the same label is used already in the existing items, the number
  * suffix is appended to the label.  This label index number is stored
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index e1cd34d9011d..0e6d7534f491 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -371,7 +371,7 @@ error:
 	return ret;
 }
 
-/**
+/*
  * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with
  * hdmi-specific routine.
  */
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index f56765ae73a7..b2d81ab22fb0 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -57,6 +57,8 @@ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 
 /**
  * snd_hda_jack_tbl_get - query the jack-table entry for the given NID
+ * @codec: the HDA codec
+ * @nid: pin NID to refer to
  */
 struct hda_jack_tbl *
 snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid)
@@ -75,6 +77,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get);
 
 /**
  * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag
+ * @codec: the HDA codec
+ * @tag: tag value to refer to
  */
 struct hda_jack_tbl *
 snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag)
@@ -93,6 +97,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag);
 
 /**
  * snd_hda_jack_tbl_new - create a jack-table entry for the given NID
+ * @codec: the HDA codec
+ * @nid: pin NID to assign
  */
 static struct hda_jack_tbl *
 snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid)
@@ -162,6 +168,7 @@ static void jack_detect_update(struct hda_codec *codec,
 
 /**
  * snd_hda_set_dirty_all - Mark all the cached as dirty
+ * @codec: the HDA codec
  *
  * This function sets the dirty flag to all entries of jack table.
  * It's called from the resume path in hda_codec.c.
@@ -218,6 +225,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
+ * @codec: the HDA codec
+ * @nid: pin NID to enable
+ * @func: callback function to register
  *
  * In the case of error, the return value will be a pointer embedded with
  * errno.  Check and handle the return value appropriately with standard
@@ -266,6 +276,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable);
 
 /**
  * snd_hda_jack_set_gating_jack - Set gating jack.
+ * @codec: the HDA codec
+ * @gated_nid: gated pin NID
+ * @gating_nid: gating pin NID
  *
  * Indicates the gated jack is only valid when the gating jack is plugged.
  */
@@ -287,6 +300,7 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack);
 
 /**
  * snd_hda_jack_report_sync - sync the states of all jacks and report if changed
+ * @codec: the HDA codec
  */
 void snd_hda_jack_report_sync(struct hda_codec *codec)
 {
@@ -349,6 +363,11 @@ static void hda_free_jack_priv(struct snd_jack *jack)
 
 /**
  * snd_hda_jack_add_kctl - Add a kctl for the given pin
+ * @codec: the HDA codec
+ * @nid: pin NID to assign
+ * @name: string name for the jack
+ * @idx: index number for the jack
+ * @phantom_jack: flag to deal as a phantom jack
  *
  * This assigns a jack-detection kctl to the given pin.  The kcontrol
  * will have the given name and index.
@@ -456,6 +475,8 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
 
 /**
  * snd_hda_jack_add_kctls - Add kctls for all pins included in the given pincfg
+ * @codec: the HDA codec
+ * @cfg: pin config table to parse
  */
 int snd_hda_jack_add_kctls(struct hda_codec *codec,
 			   const struct auto_pin_cfg *cfg)
-- 
cgit v1.2.3


From 95a962c36f6e3c3edb438d1ba59e30964900d16a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Oct 2014 16:03:58 +0100
Subject: ALSA: hda - More kerneldoc comments

Put more kerneldoc comments to the exported functions.
Still the generic parser code and the HD-audio controller code aren't
covered yet, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c |  51 +++++++++++++-
 sound/pci/hda/hda_beep.c        |  38 +++++++++++
 sound/pci/hda/hda_codec.c       | 148 ++++++++++++++++++++++++++++++++++++----
 sound/pci/hda/hda_jack.c        |  39 +++++++++++
 sound/pci/hda/hda_jack.h        |   5 ++
 sound/pci/hda/hda_sysfs.c       |  35 ++++++++--
 6 files changed, 297 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 7388958b01af..1ede82200ee5 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -441,6 +441,13 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_parse_pin_defcfg);
 
+/**
+ * snd_hda_get_input_pin_attr - Get the input pin attribute from pin config
+ * @def_conf: pin configuration value
+ *
+ * Guess the input pin attribute (INPUT_PIN_ATTR_XXX) from the given
+ * default pin configuration value.
+ */
 int snd_hda_get_input_pin_attr(unsigned int def_conf)
 {
 	unsigned int loc = get_defcfg_location(def_conf);
@@ -473,7 +480,6 @@ EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr);
  * for mic and line-in pins, and set an appropriate prefix like "Front",
  * "Rear", "Internal".
  */
-
 static const char *hda_get_input_pin_label(struct hda_codec *codec,
 					   const struct auto_pin_cfg_item *item,
 					   hda_nid_t pin, bool check_location)
@@ -761,6 +767,14 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_GPL(snd_hda_get_pin_label);
 
+/**
+ * snd_hda_add_verbs - Add verbs to the init list
+ * @codec: the HDA codec
+ * @list: zero-terminated verb list to add
+ *
+ * Append the given verb list to the execution list.  The verbs will be
+ * performed at init and resume time via snd_hda_apply_verbs().
+ */
 int snd_hda_add_verbs(struct hda_codec *codec,
 		      const struct hda_verb *list)
 {
@@ -773,6 +787,10 @@ int snd_hda_add_verbs(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_add_verbs);
 
+/**
+ * snd_hda_apply_verbs - Execute the init verb lists
+ * @codec: the HDA codec
+ */
 void snd_hda_apply_verbs(struct hda_codec *codec)
 {
 	int i;
@@ -783,6 +801,11 @@ void snd_hda_apply_verbs(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_apply_verbs);
 
+/**
+ * snd_hda_apply_pincfgs - Set each pin config in the given list
+ * @codec: the HDA codec
+ * @cfg: NULL-terminated pin config table
+ */
 void snd_hda_apply_pincfgs(struct hda_codec *codec,
 			   const struct hda_pintbl *cfg)
 {
@@ -850,6 +873,11 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth)
 	}
 }
 
+/**
+ * snd_hda_apply_fixup - Apply the fixup chain with the given action
+ * @codec: the HDA codec
+ * @action: fixup action (HDA_FIXUP_ACT_XXX)
+ */
 void snd_hda_apply_fixup(struct hda_codec *codec, int action)
 {
 	if (codec->fixup_list)
@@ -868,6 +896,12 @@ static bool pin_config_match(struct hda_codec *codec,
 	return true;
 }
 
+/**
+ * snd_hda_pick_pin_fixup - Pick up a fixup matching with the pin quirk list
+ * @codec: the HDA codec
+ * @pin_quirk: zero-terminated pin quirk list
+ * @fixlist: the fixup list
+ */
 void snd_hda_pick_pin_fixup(struct hda_codec *codec,
 			    const struct snd_hda_pin_quirk *pin_quirk,
 			    const struct hda_fixup *fixlist)
@@ -894,6 +928,21 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_pick_pin_fixup);
 
+/**
+ * snd_hda_pick_fixup - Pick up a fixup matching with PCI/codec SSID or model string
+ * @codec: the HDA codec
+ * @models: NULL-terminated model string list
+ * @quirk: zero-terminated PCI/codec SSID quirk list
+ * @fixlist: the fixup list
+ *
+ * Pick up a fixup entry matching with the given model string or SSID.
+ * If a fixup was already set beforehand, the function doesn't do anything.
+ * When a special model string "nofixup" is given, also no fixup is applied.
+ *
+ * The function tries to find the matching model name at first, if given.
+ * If nothing matched, try to look up the PCI SSID.
+ * If still nothing matched, try to look up the codec SSID.
+ */
 void snd_hda_pick_fixup(struct hda_codec *codec,
 			const struct hda_model_fixup *models,
 			const struct snd_pci_quirk *quirk,
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 8c6c50afc0b7..1e7de08e77cb 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -175,6 +175,11 @@ static int snd_hda_do_attach(struct hda_beep *beep)
 	return 0;
 }
 
+/**
+ * snd_hda_enable_beep_device - Turn on/off beep sound
+ * @codec: the HDA codec
+ * @enable: flag to turn on/off
+ */
 int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 {
 	struct hda_beep *beep = codec->beep;
@@ -191,6 +196,20 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 }
 EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device);
 
+/**
+ * snd_hda_attach_beep_device - Attach a beep input device
+ * @codec: the HDA codec
+ * @nid: beep NID
+ *
+ * Attach a beep object to the given widget.  If beep hint is turned off
+ * explicitly or beep_mode of the codec is turned off, this doesn't nothing.
+ *
+ * The attached beep device has to be registered via
+ * snd_hda_register_beep_device() and released via snd_hda_detach_beep_device()
+ * appropriately.
+ *
+ * Currently, only one beep device is allowed to each codec.
+ */
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 {
 	struct hda_beep *beep;
@@ -228,6 +247,10 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 }
 EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device);
 
+/**
+ * snd_hda_detach_beep_device - Detach the beep device
+ * @codec: the HDA codec
+ */
 void snd_hda_detach_beep_device(struct hda_codec *codec)
 {
 	struct hda_beep *beep = codec->beep;
@@ -240,6 +263,10 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device);
 
+/**
+ * snd_hda_register_beep_device - Register the beep device
+ * @codec: the HDA codec
+ */
 int snd_hda_register_beep_device(struct hda_codec *codec)
 {
 	struct hda_beep *beep = codec->beep;
@@ -269,6 +296,12 @@ static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
 }
 
 /* get/put callbacks for beep mute mixer switches */
+
+/**
+ * snd_hda_mixer_amp_switch_get_beep - Get callback for beep controls
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
+ */
 int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
@@ -283,6 +316,11 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get_beep);
 
+/**
+ * snd_hda_mixer_amp_switch_put_beep - Put callback for beep controls
+ * @kcontrol: ctl element
+ * @ucontrol: pointer to get/store the data
+ */
 int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e15254204c72..ca98f5209f8f 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -77,6 +77,10 @@ static struct hda_vendor_id hda_vendor_ids[] = {
 static DEFINE_MUTEX(preset_mutex);
 static LIST_HEAD(hda_preset_tables);
 
+/**
+ * snd_hda_add_codec_preset - Add a codec preset to the chain
+ * @preset: codec preset table to add
+ */
 int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset)
 {
 	mutex_lock(&preset_mutex);
@@ -86,6 +90,10 @@ int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset)
 }
 EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset);
 
+/**
+ * snd_hda_delete_codec_preset - Delete a codec preset from the chain
+ * @preset: codec preset table to delete
+ */
 int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset)
 {
 	mutex_lock(&preset_mutex);
@@ -1187,7 +1195,16 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_get_pincfg);
 
-/* remember the current pinctl target value */
+/**
+ * snd_hda_codec_set_pin_target - remember the current pinctl target value
+ * @codec: the HDA codec
+ * @nid: pin NID
+ * @val: assigned pinctl value
+ *
+ * This function stores the given value to a pinctl target value in the
+ * pincfg table.  This isn't always as same as the actually written value
+ * but can be referred at any time via snd_hda_codec_get_pin_target().
+ */
 int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid,
 				 unsigned int val)
 {
@@ -1201,7 +1218,11 @@ int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_set_pin_target);
 
-/* return the current pinctl target value */
+/**
+ * snd_hda_codec_get_pin_target - return the current pinctl target value
+ * @codec: the HDA codec
+ * @nid: pin NID
+ */
 int snd_hda_codec_get_pin_target(struct hda_codec *codec, hda_nid_t nid)
 {
 	struct hda_pincfg *pin;
@@ -1573,6 +1594,13 @@ int snd_hda_codec_new(struct hda_bus *bus,
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_new);
 
+/**
+ * snd_hda_codec_update_widgets - Refresh widget caps and pin defaults
+ * @codec: the HDA codec
+ *
+ * Forcibly refresh the all widget caps and the init pin configurations of
+ * the given codec.
+ */
 int snd_hda_codec_update_widgets(struct hda_codec *codec)
 {
 	hda_nid_t fg;
@@ -2239,7 +2267,17 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo);
 
-/* Works like snd_hda_codec_amp_update() but it writes the value only at
+/**
+ * snd_hda_codec_amp_init - initialize the AMP value
+ * @codec: the HDA codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @dir: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Works like snd_hda_codec_amp_update() but it writes the value only at
  * the first access.  If the amp was already initialized / updated beforehand,
  * this does nothing.
  */
@@ -2250,6 +2288,17 @@ int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init);
 
+/**
+ * snd_hda_codec_amp_init_stereo - initialize the stereo AMP value
+ * @codec: the HDA codec
+ * @nid: NID to read the AMP value
+ * @dir: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Call snd_hda_codec_amp_init() for both stereo channels.
+ */
 int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid,
 				  int dir, int idx, int mask, int val)
 {
@@ -2644,7 +2693,10 @@ void snd_hda_ctls_clear(struct hda_codec *codec)
 	snd_array_free(&codec->nids);
 }
 
-/* pseudo device locking
+/**
+ * snd_hda_lock_devices - pseudo device locking
+ * @bus: the BUS
+ *
  * toggle card->shutdown to allow/disallow the device access (as a hack)
  */
 int snd_hda_lock_devices(struct hda_bus *bus)
@@ -2681,6 +2733,10 @@ int snd_hda_lock_devices(struct hda_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hda_lock_devices);
 
+/**
+ * snd_hda_unlock_devices - pseudo device unlocking
+ * @bus: the BUS
+ */
 void snd_hda_unlock_devices(struct hda_bus *bus)
 {
 	struct snd_card *card = bus->card;
@@ -2867,7 +2923,7 @@ static int add_slave(struct hda_codec *codec,
 }
 
 /**
- * snd_hda_add_vmaster - create a virtual master control and add slaves
+ * __snd_hda_add_vmaster - create a virtual master control and add slaves
  * @codec: HD-audio codec
  * @name: vmaster control name
  * @tlv: TLV data (optional)
@@ -2970,10 +3026,15 @@ static struct snd_kcontrol_new vmaster_mute_mode = {
 	.put = vmaster_mute_mode_put,
 };
 
-/*
- * Add a mute-LED hook with the given vmaster switch kctl
- * "Mute-LED Mode" control is automatically created and associated with
- * the given hook.
+/**
+ * snd_hda_add_vmaster_hook - Add a vmaster hook for mute-LED
+ * @codec: the HDA codec
+ * @hook: the vmaster hook object
+ * @expose_enum_ctl: flag to create an enum ctl
+ *
+ * Add a mute-LED hook with the given vmaster switch kctl.
+ * When @expose_enum_ctl is set, "Mute-LED Mode" control is automatically
+ * created and associated with the given hook.
  */
 int snd_hda_add_vmaster_hook(struct hda_codec *codec,
 			     struct hda_vmaster_mute_hook *hook,
@@ -2995,9 +3056,12 @@ int snd_hda_add_vmaster_hook(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_add_vmaster_hook);
 
-/*
- * Call the hook with the current value for synchronization
- * Should be called in init callback
+/**
+ * snd_hda_sync_vmaster_hook - Sync vmaster hook
+ * @hook: the vmaster hook
+ *
+ * Call the hook with the current value for synchronization.
+ * Should be called in init callback.
  */
 void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook)
 {
@@ -3599,7 +3663,11 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls);
 
-/* get the hda_spdif_out entry from the given NID
+/**
+ * snd_hda_spdif_out_of_nid - get the hda_spdif_out entry from the given NID
+ * @codec: the HDA codec
+ * @nid: widget NID
+ *
  * call within spdif_mutex lock
  */
 struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
@@ -3616,6 +3684,13 @@ struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_spdif_out_of_nid);
 
+/**
+ * snd_hda_spdif_ctls_unassign - Unassign the given SPDIF ctl
+ * @codec: the HDA codec
+ * @idx: the SPDIF ctl index
+ *
+ * Unassign the widget from the given SPDIF control.
+ */
 void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
 {
 	struct hda_spdif_out *spdif;
@@ -3627,6 +3702,14 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
 }
 EXPORT_SYMBOL_GPL(snd_hda_spdif_ctls_unassign);
 
+/**
+ * snd_hda_spdif_ctls_assign - Assign the SPDIF controls to the given NID
+ * @codec: the HDA codec
+ * @idx: the SPDIF ctl idx
+ * @nid: widget NID
+ *
+ * Assign the widget to the SPDIF control with the given index.
+ */
 void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
 {
 	struct hda_spdif_out *spdif;
@@ -3946,6 +4029,16 @@ void snd_hda_codec_flush_cache(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_flush_cache);
 
+/**
+ * snd_hda_codec_set_power_to_all - Set the power state to all widgets
+ * @codec: the HDA codec
+ * @fg: function group (not used now)
+ * @power_state: the power state to set (AC_PWRST_*)
+ *
+ * Set the given power state to all widgets that have the power control.
+ * If the codec has power_filter set, it evaluates the power state and
+ * filter out if it's unchanged as D3.
+ */
 void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
 				    unsigned int power_state)
 {
@@ -4010,7 +4103,15 @@ static unsigned int hda_sync_power_state(struct hda_codec *codec,
 	return state;
 }
 
-/* don't power down the widget if it controls eapd and EAPD_BTLENABLE is set */
+/**
+ * snd_hda_codec_eapd_power_filter - A power filter callback for EAPD
+ * @codec: the HDA codec
+ * @nid: widget NID
+ * @power_state: power state to evalue
+ *
+ * Don't power down the widget if it controls eapd and EAPD_BTLENABLE is set.
+ * This can be used a codec power_filter callback.
+ */
 unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
 					     hda_nid_t nid,
 					     unsigned int power_state)
@@ -4671,6 +4772,17 @@ static int set_pcm_default_values(struct hda_codec *codec,
 /*
  * codec prepare/cleanup entries
  */
+/**
+ * snd_hda_codec_prepare - Prepare a stream
+ * @codec: the HDA codec
+ * @hinfo: PCM information
+ * @stream: stream tag to assign
+ * @format: format id to assign
+ * @substream: PCM substream to assign
+ *
+ * Calls the prepare callback set by the codec with the given arguments.
+ * Clean up the inactive streams when successful.
+ */
 int snd_hda_codec_prepare(struct hda_codec *codec,
 			  struct hda_pcm_stream *hinfo,
 			  unsigned int stream,
@@ -4687,6 +4799,14 @@ int snd_hda_codec_prepare(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_prepare);
 
+/**
+ * snd_hda_codec_cleanup - Prepare a stream
+ * @codec: the HDA codec
+ * @hinfo: PCM information
+ * @substream: PCM substream
+ *
+ * Calls the cleanup callback set by the codec with the given arguments.
+ */
 void snd_hda_codec_cleanup(struct hda_codec *codec,
 			   struct hda_pcm_stream *hinfo,
 			   struct snd_pcm_substream *substream)
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index b2d81ab22fb0..e664307617bd 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -20,6 +20,16 @@
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
 
+/**
+ * is_jack_detectable - Check whether the given pin is jack-detectable
+ * @codec: the HDA codec
+ * @nid: pin NID
+ *
+ * Check whether the given pin is capable to report the jack detection.
+ * The jack detection might not work by various reasons, e.g. the jack
+ * detection is prohibited in the codec level, the pin config has
+ * AC_DEFCFG_MISC_NO_PRESENCE bit, no unsol support, etc.
+ */
 bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
 {
 	if (codec->no_jack_detect)
@@ -268,6 +278,14 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback);
 
+/**
+ * snd_hda_jack_detect_enable - Enable the jack detection on the given pin
+ * @codec: the HDA codec
+ * @nid: pin NID to enable jack detection
+ *
+ * Enable the jack detection with the default callback.  Returns zero if
+ * successful or a negative error code.
+ */
 int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid)
 {
 	return PTR_ERR_OR_ZERO(snd_hda_jack_detect_enable_callback(codec, nid, NULL));
@@ -410,6 +428,15 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 	return 0;
 }
 
+/**
+ * snd_hda_jack_add_kctl - Add a jack kctl for the given pin
+ * @codec: the HDA codec
+ * @nid: pin NID
+ * @name: the name string for the jack ctl
+ * @idx: the ctl index for the jack ctl
+ *
+ * This is a simple helper calling __snd_hda_jack_add_kctl().
+ */
 int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 			  const char *name, int idx)
 {
@@ -552,6 +579,11 @@ static void call_jack_callback(struct hda_codec *codec,
 	}
 }
 
+/**
+ * snd_hda_jack_unsol_event - Handle an unsolicited event
+ * @codec: the HDA codec
+ * @res: the unsolicited event data
+ */
 void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res)
 {
 	struct hda_jack_tbl *event;
@@ -567,6 +599,13 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res)
 }
 EXPORT_SYMBOL_GPL(snd_hda_jack_unsol_event);
 
+/**
+ * snd_hda_jack_poll_all - Poll all jacks
+ * @codec: the HDA codec
+ *
+ * Poll all detectable jacks with dirty flag, update the status, call
+ * callbacks and call snd_hda_jack_report_sync() if any changes are found.
+ */
 void snd_hda_jack_poll_all(struct hda_codec *codec)
 {
 	struct hda_jack_tbl *jack = codec->jacktbl.list;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 13cb375454f6..b279e327a23b 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -72,6 +72,11 @@ enum {
 
 int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
 
+/**
+ * snd_hda_jack_detect - Detect the jack
+ * @codec: the HDA codec
+ * @nid: pin NID to check jack detection
+ */
 static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
 {
 	return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 9b49f156a12e..bef721592c3a 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -417,8 +417,13 @@ static DEVICE_ATTR_RW(user_pin_configs);
 static DEVICE_ATTR_WO(reconfig);
 static DEVICE_ATTR_WO(clear);
 
-/*
- * Look for hint string
+/**
+ * snd_hda_get_hint - Look for hint string
+ * @codec: the HDA codec
+ * @key: the hint key string
+ *
+ * Look for a hint key/value pair matching with the given key string
+ * and returns the value string.  If nothing found, returns NULL.
  */
 const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
 {
@@ -427,6 +432,15 @@ const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
 }
 EXPORT_SYMBOL_GPL(snd_hda_get_hint);
 
+/**
+ * snd_hda_get_bool_hint - Get a boolean hint value
+ * @codec: the HDA codec
+ * @key: the hint key string
+ *
+ * Look for a hint key/value pair matching with the given key string
+ * and returns a boolean value parsed from the value.  If no matching
+ * key is found, return a negative value.
+ */
 int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 {
 	const char *p;
@@ -453,6 +467,16 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 }
 EXPORT_SYMBOL_GPL(snd_hda_get_bool_hint);
 
+/**
+ * snd_hda_get_bool_hint - Get a boolean hint value
+ * @codec: the HDA codec
+ * @key: the hint key string
+ * @valp: pointer to store a value
+ *
+ * Look for a hint key/value pair matching with the given key string
+ * and stores the integer value to @valp.  If no matching key is found,
+ * return a negative error code.  Otherwise it returns zero.
+ */
 int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 {
 	const char *p;
@@ -690,8 +714,11 @@ static int get_line_from_fw(char *buf, int size, size_t *fw_size_p,
 	return 1;
 }
 
-/*
- * load a "patch" firmware file and parse it
+/**
+ * snd_hda_load_patch - load a "patch" firmware file and parse it
+ * @bus: HD-audio bus
+ * @fw_size: the firmware byte size
+ * @fw_buf: the firmware data
  */
 int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf)
 {
-- 
cgit v1.2.3


From df57de172a47f16548ee4bb69d1110e32686d6a9 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Wed, 29 Oct 2014 20:09:45 +0530
Subject: ALSA: hdspm: remove unused variable

removed the unused variables. These variables were only being
assigned some value, but the values were never being used.

it has been build tested after removing the variables.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdspm.c | 23 +++--------------------
 1 file changed, 3 insertions(+), 20 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 7f7277bfb66a..e09348c156d8 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1257,14 +1257,13 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate)
 /* check for external sample rate, returns the sample rate in Hz*/
 static int hdspm_external_sample_rate(struct hdspm *hdspm)
 {
-	unsigned int status, status2, timecode;
+	unsigned int status, status2;
 	int syncref, rate = 0, rate_bits;
 
 	switch (hdspm->io_type) {
 	case AES32:
 		status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 		status = hdspm_read(hdspm, HDSPM_statusRegister);
-		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
 		syncref = hdspm_autosync_ref(hdspm);
 		switch (syncref) {
@@ -4862,18 +4861,15 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
 			 struct snd_info_buffer *buffer)
 {
 	struct hdspm *hdspm = entry->private_data;
-	unsigned int status, status2, control, freq;
+	unsigned int status, status2;
 
 	char *pref_sync_ref;
 	char *autosync_ref;
 	char *system_clock_mode;
-	char *insel;
 	int x, x2;
 
 	status = hdspm_read(hdspm, HDSPM_statusRegister);
 	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
-	control = hdspm->control_register;
-	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
 	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
 			hdspm->card_name, hdspm->card->number + 1,
@@ -4936,17 +4932,6 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "Line out: %s\n",
 		(hdspm->control_register & HDSPM_LineOut) ? "on " : "off");
 
-	switch (hdspm->control_register & HDSPM_InputMask) {
-	case HDSPM_InputOptical:
-		insel = "Optical";
-		break;
-	case HDSPM_InputCoaxial:
-		insel = "Coaxial";
-		break;
-	default:
-		insel = "Unknown";
-	}
-
 	snd_iprintf(buffer,
 		"ClearTrackMarker = %s, Transmit in %s Channel Mode, "
 		"Auto Input %s\n",
@@ -5191,15 +5176,13 @@ snd_hdspm_proc_read_raydat(struct snd_info_entry *entry,
 			 struct snd_info_buffer *buffer)
 {
 	struct hdspm *hdspm = entry->private_data;
-	unsigned int status1, status2, status3, control, i;
+	unsigned int status1, status2, status3, i;
 	unsigned int lock, sync;
 
 	status1 = hdspm_read(hdspm, HDSPM_RD_STATUS_1); /* s1 */
 	status2 = hdspm_read(hdspm, HDSPM_RD_STATUS_2); /* freq */
 	status3 = hdspm_read(hdspm, HDSPM_RD_STATUS_3); /* s2 */
 
-	control = hdspm->control_register;
-
 	snd_iprintf(buffer, "STATUS1: 0x%08x\n", status1);
 	snd_iprintf(buffer, "STATUS2: 0x%08x\n", status2);
 	snd_iprintf(buffer, "STATUS3: 0x%08x\n", status3);
-- 
cgit v1.2.3


From eafe8404c103b3051b6421fc17e0e8b91d369f0b Mon Sep 17 00:00:00 2001
From: Tina Ruchandani <ruchandani.tina@gmail.com>
Date: Wed, 29 Oct 2014 10:48:10 -0700
Subject: ALSA: es1968: Replace timeval with ktime_t

es1968_measure_clock uses struct timeval, which on 32-bit systems will overflow
in 2038, leading to incorrect interpretation of time.This patch changes the
function to use ktime_t instead of struct timeval, which implies:
- no y2038: ktime_t uses a 64-bit datatype explicitly.
- efficent subtraction: The earlier version computes the difference in usecs
  while dealing with secs and nsecs. It requires checks to see if the nsecs of
  stop is less than start. This patch uses a direct subtract of ktime_t and
  converts to usecs.
- use of monotonic clock (ktime_get) over real time (do_gettimeofday),
  which simplifies timekeeping, as it does not have to deal with cases
  where stop_time is less than start_time.

Signed-off-by: Tina Ruchandani <ruchandani.tina@gmail.com>
Reviewed-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/es1968.c | 15 ++++++---------
 1 file changed, 6 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index a9956a7c5677..6039700f8579 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -1710,7 +1710,8 @@ static void es1968_measure_clock(struct es1968 *chip)
 	int i, apu;
 	unsigned int pa, offset, t;
 	struct esm_memory *memory;
-	struct timeval start_time, stop_time;
+	ktime_t start_time, stop_time;
+	ktime_t diff;
 
 	if (chip->clock == 0)
 		chip->clock = 48000; /* default clock value */
@@ -1761,12 +1762,12 @@ static void es1968_measure_clock(struct es1968 *chip)
 	snd_es1968_bob_inc(chip, ESM_BOB_FREQ);
 	__apu_set_register(chip, apu, 5, pa & 0xffff);
 	snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR);
-	do_gettimeofday(&start_time);
+	start_time = ktime_get();
 	spin_unlock_irq(&chip->reg_lock);
 	msleep(50);
 	spin_lock_irq(&chip->reg_lock);
 	offset = __apu_get_register(chip, apu, 5);
-	do_gettimeofday(&stop_time);
+	stop_time = ktime_get();
 	snd_es1968_trigger_apu(chip, apu, 0); /* stop */
 	snd_es1968_bob_dec(chip);
 	chip->in_measurement = 0;
@@ -1777,12 +1778,8 @@ static void es1968_measure_clock(struct es1968 *chip)
 	offset &= 0xfffe;
 	offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2);
 
-	t = stop_time.tv_sec - start_time.tv_sec;
-	t *= 1000000;
-	if (stop_time.tv_usec < start_time.tv_usec)
-		t -= start_time.tv_usec - stop_time.tv_usec;
-	else
-		t += stop_time.tv_usec - start_time.tv_usec;
+	diff = ktime_sub(stop_time, start_time);
+	t = ktime_to_us(diff);
 	if (t == 0) {
 		dev_err(chip->card->dev, "?? calculation error..\n");
 	} else {
-- 
cgit v1.2.3


From dda42bd0c3a4b7be1561546914eda59b68a58be4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Oct 2014 12:04:50 +0100
Subject: ALSA: hda - Add kerneldoc comments to hda_generic.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 164 ++++++++++++++++++++++++++++++++++++--------
 1 file changed, 135 insertions(+), 29 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 06d721085e72..63b69f750d8e 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -40,7 +40,12 @@
 #include "hda_generic.h"
 
 
-/* initialize hda_gen_spec struct */
+/**
+ * snd_hda_gen_spec_init - initialize hda_gen_spec struct
+ * @spec: hda_gen_spec object to initialize
+ *
+ * Initialize the given hda_gen_spec object.
+ */
 int snd_hda_gen_spec_init(struct hda_gen_spec *spec)
 {
 	snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32);
@@ -51,6 +56,17 @@ int snd_hda_gen_spec_init(struct hda_gen_spec *spec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_spec_init);
 
+/**
+ * snd_hda_gen_add_kctl - Add a new kctl_new struct from the template
+ * @spec: hda_gen_spec object
+ * @name: name string to override the template, NULL if unchanged
+ * @temp: template for the new kctl
+ *
+ * Add a new kctl (actually snd_kcontrol_new to be instantiated later)
+ * element based on the given snd_kcontrol_new template @temp and the
+ * name string @name to the list in @spec.
+ * Returns the newly created object or NULL as error.
+ */
 struct snd_kcontrol_new *
 snd_hda_gen_add_kctl(struct hda_gen_spec *spec, const char *name,
 		     const struct snd_kcontrol_new *temp)
@@ -259,8 +275,14 @@ static struct nid_path *get_nid_path(struct hda_codec *codec,
 	return NULL;
 }
 
-/* get the path between the given NIDs;
- * passing 0 to either @pin or @dac behaves as a wildcard
+/**
+ * snd_hda_get_nid_path - get the path between the given NIDs
+ * @codec: the HDA codec
+ * @from_nid: the NID where the path start from
+ * @to_nid: the NID where the path ends at
+ *
+ * Return the found nid_path object or NULL for error.
+ * Passing 0 to either @from_nid or @to_nid behaves as a wildcard.
  */
 struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec,
 				      hda_nid_t from_nid, hda_nid_t to_nid)
@@ -269,8 +291,14 @@ struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_get_nid_path);
 
-/* get the index number corresponding to the path instance;
- * the index starts from 1, for easier checking the invalid value
+/**
+ * snd_hda_get_path_idx - get the index number corresponding to the path
+ * instance
+ * @codec: the HDA codec
+ * @path: nid_path object
+ *
+ * The returned index starts from 1, i.e. the actual array index with offset 1,
+ * and zero is handled as an invalid path
  */
 int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path)
 {
@@ -287,7 +315,12 @@ int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path)
 }
 EXPORT_SYMBOL_GPL(snd_hda_get_path_idx);
 
-/* get the path instance corresponding to the given index number */
+/**
+ * snd_hda_get_path_from_idx - get the path instance corresponding to the
+ * given index number
+ * @codec: the HDA codec
+ * @idx: the path index
+ */
 struct nid_path *snd_hda_get_path_from_idx(struct hda_codec *codec, int idx)
 {
 	struct hda_gen_spec *spec = codec->spec;
@@ -415,7 +448,18 @@ static bool __parse_nid_path(struct hda_codec *codec,
 	return true;
 }
 
-/* parse the widget path from the given nid to the target nid;
+/**
+ * snd_hda_parse_nid_path - parse the widget path from the given nid to
+ * the target nid
+ * @codec: the HDA codec
+ * @from_nid: the NID where the path start from
+ * @to_nid: the NID where the path ends at
+ * @anchor_nid: the anchor indication
+ * @path: the path object to store the result
+ *
+ * Returns true if a matching path is found.
+ *
+ * The parsing behavior depends on parameters:
  * when @from_nid is 0, try to find an empty DAC;
  * when @anchor_nid is set to a positive value, only paths through the widget
  * with the given value are evaluated.
@@ -436,9 +480,15 @@ bool snd_hda_parse_nid_path(struct hda_codec *codec, hda_nid_t from_nid,
 }
 EXPORT_SYMBOL_GPL(snd_hda_parse_nid_path);
 
-/*
- * parse the path between the given NIDs and add to the path list.
- * if no valid path is found, return NULL
+/**
+ * snd_hda_add_new_path - parse the path between the given NIDs and
+ * add to the path list
+ * @codec: the HDA codec
+ * @from_nid: the NID where the path start from
+ * @to_nid: the NID where the path ends at
+ * @anchor_nid: the anchor indication, see snd_hda_parse_nid_path()
+ *
+ * If no valid path is found, returns NULL.
  */
 struct nid_path *
 snd_hda_add_new_path(struct hda_codec *codec, hda_nid_t from_nid,
@@ -724,8 +774,14 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path,
 	}
 }
 
-/* activate or deactivate the given path
- * if @add_aamix is set, enable the input from aa-mix NID as well (if any)
+/**
+ * snd_hda_activate_path - activate or deactivate the given path
+ * @codec: the HDA codec
+ * @path: the path to activate/deactivate
+ * @enable: flag to activate or not
+ * @add_aamix: enable the input from aamix NID
+ *
+ * If @add_aamix is set, enable the input from aa-mix NID as well (if any).
  */
 void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path,
 			   bool enable, bool add_aamix)
@@ -3883,7 +3939,12 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 	}
 }
 
-/* Toggle outputs muting */
+/**
+ * snd_hda_gen_update_outputs - Toggle outputs muting
+ * @codec: the HDA codec
+ *
+ * Update the mute status of all outputs based on the current jack states.
+ */
 void snd_hda_gen_update_outputs(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
@@ -3944,7 +4005,11 @@ static void call_update_outputs(struct hda_codec *codec)
 		snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false);
 }
 
-/* standard HP-automute helper */
+/**
+ * snd_hda_gen_hp_automute - standard HP-automute helper
+ * @codec: the HDA codec
+ * @jack: jack object, NULL for the whole
+ */
 void snd_hda_gen_hp_automute(struct hda_codec *codec,
 			     struct hda_jack_callback *jack)
 {
@@ -3965,7 +4030,11 @@ void snd_hda_gen_hp_automute(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_hp_automute);
 
-/* standard line-out-automute helper */
+/**
+ * snd_hda_gen_line_automute - standard line-out-automute helper
+ * @codec: the HDA codec
+ * @jack: jack object, NULL for the whole
+ */
 void snd_hda_gen_line_automute(struct hda_codec *codec,
 			       struct hda_jack_callback *jack)
 {
@@ -3986,7 +4055,11 @@ void snd_hda_gen_line_automute(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_line_automute);
 
-/* standard mic auto-switch helper */
+/**
+ * snd_hda_gen_mic_autoswitch - standard mic auto-switch helper
+ * @codec: the HDA codec
+ * @jack: jack object, NULL for the whole
+ */
 void snd_hda_gen_mic_autoswitch(struct hda_codec *codec,
 				struct hda_jack_callback *jack)
 {
@@ -4318,7 +4391,13 @@ static int check_auto_mic_availability(struct hda_codec *codec)
 	return 0;
 }
 
-/* power_filter hook; make inactive widgets into power down */
+/**
+ * snd_hda_gen_path_power_filter - power_filter hook to make inactive widgets
+ * into power down
+ * @codec: the HDA codec
+ * @nid: NID to evalute
+ * @power_state: target power state
+ */
 unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 						  hda_nid_t nid,
 						  unsigned int power_state)
@@ -4354,8 +4433,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
 	}
 }
 
-/*
- * Parse the given BIOS configuration and set up the hda_gen_spec
+/**
+ * snd_hda_gen_parse_auto_config - Parse the given BIOS configuration and
+ * set up the hda_gen_spec
+ * @codec: the HDA codec
+ * @cfg: Parsed pin configuration
  *
  * return 1 if successful, 0 if the proper config is not found,
  * or a negative error code
@@ -4541,6 +4623,12 @@ static const char * const slave_pfxs[] = {
 	NULL,
 };
 
+/**
+ * snd_hda_gen_build_controls - Build controls from the parsed results
+ * @codec: the HDA codec
+ *
+ * Pass this to build_controls patch_ops.
+ */
 int snd_hda_gen_build_controls(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
@@ -5018,7 +5106,12 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx,
 	strlcat(str, sfx, len);
 }
 
-/* build PCM streams based on the parsed results */
+/**
+ * snd_hda_gen_build_pcms - build PCM streams based on the parsed results
+ * @codec: the HDA codec
+ *
+ * Pass this to build_pcms patch_ops.
+ */
 int snd_hda_gen_build_pcms(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
@@ -5313,9 +5406,11 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec)
 	}
 }
 
-/*
- * initialize the generic spec;
- * this can be put as patch_ops.init function
+/**
+ * snd_hda_gen_init - initialize the generic spec
+ * @codec: the HDA codec
+ *
+ * This can be put as patch_ops init function.
  */
 int snd_hda_gen_init(struct hda_codec *codec)
 {
@@ -5351,9 +5446,11 @@ int snd_hda_gen_init(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_init);
 
-/*
- * free the generic spec;
- * this can be put as patch_ops.free function
+/**
+ * snd_hda_gen_free - free the generic spec
+ * @codec: the HDA codec
+ *
+ * This can be put as patch_ops free function.
  */
 void snd_hda_gen_free(struct hda_codec *codec)
 {
@@ -5365,9 +5462,12 @@ void snd_hda_gen_free(struct hda_codec *codec)
 EXPORT_SYMBOL_GPL(snd_hda_gen_free);
 
 #ifdef CONFIG_PM
-/*
- * check the loopback power save state;
- * this can be put as patch_ops.check_power_status function
+/**
+ * snd_hda_gen_check_power_status - check the loopback power save state
+ * @codec: the HDA codec
+ * @nid: NID to inspect
+ *
+ * This can be put as patch_ops check_power_status function.
  */
 int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 {
@@ -5393,6 +5493,12 @@ static const struct hda_codec_ops generic_patch_ops = {
 #endif
 };
 
+/**
+ * snd_hda_parse_generic_codec - Generic codec parser
+ * @codec: the HDA codec
+ *
+ * This should be called from the HDA codec core.
+ */
 int snd_hda_parse_generic_codec(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec;
-- 
cgit v1.2.3


From e369086968157415aeb11af3b57cd998c6721603 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Mon, 3 Nov 2014 16:04:12 +0530
Subject: ALSA: echoaudio: add reference of struct echoaudio

added reference of struct echoaudio to free_firmware function.
this structure will be later used to get a reference of the card
when converting snd_printk to dev_* in the next patch of the series.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c     |  3 ++-
 sound/pci/echoaudio/echoaudio.h     |  3 ++-
 sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++-----
 3 files changed, 9 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index d82321ff549b..db1b247d8587 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry,
 
 
 
-static void free_firmware(const struct firmware *fw_entry)
+static void free_firmware(const struct firmware *fw_entry,
+			  struct echoaudio *chip)
 {
 #ifdef CONFIG_PM_SLEEP
 	DE_ACT(("firmware not released (kept in cache)\n"));
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index b86b88da81cd..a4f112aa78e2 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip);
 static int send_vector(struct echoaudio *chip, u32 command);
 static int get_firmware(const struct firmware **fw_entry,
 			struct echoaudio *chip, const short fw_index);
-static void free_firmware(const struct firmware *fw_entry);
+static void free_firmware(const struct firmware *fw_entry,
+			  struct echoaudio *chip);
 
 #ifdef ECHOCARD_HAS_MIDI
 static int enable_midi_input(struct echoaudio *chip, char enable);
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 5a6a217b82e0..977b2bd2e72f 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic)
 	}
 
 	DE_INIT(("ASIC loaded\n"));
-	free_firmware(fw);
+	free_firmware(fw, chip);
 	return 0;
 
 la_error:
 	DE_INIT(("failed on write_dsp\n"));
-	free_firmware(fw);
+	free_firmware(fw, chip);
 	return -EIO;
 }
 
@@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip)
 	}
 
 	DE_INIT(("Resident loader successfully installed\n"));
-	free_firmware(fw);
+	free_firmware(fw, chip);
 	return 0;
 
 irl_error:
-	free_firmware(fw);
+	free_firmware(fw, chip);
 	return -EIO;
 }
 
@@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip)
 	if (err < 0)
 		return err;
 	err = load_dsp(chip, (u16 *)fw->data);
-	free_firmware(fw);
+	free_firmware(fw, chip);
 	if (err < 0)
 		return err;
 
-- 
cgit v1.2.3


From b5b4a41b392960010fccf1f9ccf8334d612bd450 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Mon, 3 Nov 2014 16:04:13 +0530
Subject: ALSA: echoaudio: remove all snd_printk

removed all references of snd_printk with the standard dev_* macro.

[a few places degraded to dev_dbg(), too -- tiwai]

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/darla20_dsp.c        |   7 +-
 sound/pci/echoaudio/darla24_dsp.c        |  15 ++--
 sound/pci/echoaudio/echo3g_dsp.c         |   7 +-
 sound/pci/echoaudio/echoaudio.c          | 137 ++++++++++++++++---------------
 sound/pci/echoaudio/echoaudio.h          |  28 -------
 sound/pci/echoaudio/echoaudio_3g.c       |  32 ++++----
 sound/pci/echoaudio/echoaudio_dsp.c      | 111 +++++++++++++++----------
 sound/pci/echoaudio/echoaudio_gml.c      |  11 +--
 sound/pci/echoaudio/gina20_dsp.c         |  15 ++--
 sound/pci/echoaudio/gina24_dsp.c         |  39 +++++----
 sound/pci/echoaudio/indigo_dsp.c         |  13 +--
 sound/pci/echoaudio/indigo_express_dsp.c |   6 +-
 sound/pci/echoaudio/indigodj_dsp.c       |  13 +--
 sound/pci/echoaudio/indigodjx_dsp.c      |   7 +-
 sound/pci/echoaudio/indigoio_dsp.c       |  10 ++-
 sound/pci/echoaudio/indigoiox_dsp.c      |   7 +-
 sound/pci/echoaudio/layla20_dsp.c        |  37 +++++----
 sound/pci/echoaudio/layla24_dsp.c        |  34 ++++----
 sound/pci/echoaudio/mia_dsp.c            |  17 ++--
 sound/pci/echoaudio/midi.c               |  29 +++----
 sound/pci/echoaudio/mona_dsp.c           |  35 ++++----
 21 files changed, 327 insertions(+), 283 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c
index 20c7cbc89bb3..c94e92e31ae6 100644
--- a/sound/pci/echoaudio/darla20_dsp.c
+++ b/sound/pci/echoaudio/darla20_dsp.c
@@ -33,12 +33,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Darla20\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Darla20\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw: could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -57,7 +58,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw: done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c
index 6da6663e9176..b1272f88d59d 100644
--- a/sound/pci/echoaudio/darla24_dsp.c
+++ b/sound/pci/echoaudio/darla24_dsp.c
@@ -33,12 +33,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Darla24\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Darla24\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw: could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -56,7 +57,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw: done\n");
 	return err;
 }
 
@@ -128,15 +129,17 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 		clock = GD24_8000;
 		break;
 	default:
-		DE_ACT(("set_sample_rate: Error, invalid sample rate %d\n",
-			rate));
+		dev_err(chip->card->dev,
+			"set_sample_rate: Error, invalid sample rate %d\n",
+			rate);
 		return -EINVAL;
 	}
 
 	if (wait_handshake(chip))
 		return -EIO;
 
-	DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+	dev_dbg(chip->card->dev,
+		"set_sample_rate: %d clock %d\n", rate, clock);
 	chip->sample_rate = rate;
 
 	/* Override the sample rate if this card is set to Echo sync. */
diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
index 3cdc2ee2d1dd..bc3716895fc8 100644
--- a/sound/pci/echoaudio/echo3g_dsp.c
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -46,12 +46,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	int err;
 
 	local_irq_enable();
-	DE_INIT(("init_hw() - Echo3G\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Echo3G\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -98,7 +99,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 				ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
 				ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index db1b247d8587..1ef29e5b53a7 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -48,13 +48,16 @@ static int get_firmware(const struct firmware **fw_entry,
 
 #ifdef CONFIG_PM_SLEEP
 	if (chip->fw_cache[fw_index]) {
-		DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data));
+		dev_dbg(chip->card->dev,
+			"firmware requested: %s is cached\n",
+			card_fw[fw_index].data);
 		*fw_entry = chip->fw_cache[fw_index];
 		return 0;
 	}
 #endif
 
-	DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data));
+	dev_dbg(chip->card->dev,
+		"firmware requested: %s\n", card_fw[fw_index].data);
 	snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data);
 	err = request_firmware(fw_entry, name, pci_device(chip));
 	if (err < 0)
@@ -73,10 +76,10 @@ static void free_firmware(const struct firmware *fw_entry,
 			  struct echoaudio *chip)
 {
 #ifdef CONFIG_PM_SLEEP
-	DE_ACT(("firmware not released (kept in cache)\n"));
+	dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n");
 #else
 	release_firmware(fw_entry);
-	DE_ACT(("firmware released\n"));
+	dev_dbg(chip->card->dev, "firmware released\n");
 #endif
 }
 
@@ -90,10 +93,10 @@ static void free_firmware_cache(struct echoaudio *chip)
 	for (i = 0; i < 8 ; i++)
 		if (chip->fw_cache[i]) {
 			release_firmware(chip->fw_cache[i]);
-			DE_ACT(("release_firmware(%d)\n", i));
+			dev_dbg(chip->card->dev, "release_firmware(%d)\n", i);
 		}
 
-	DE_ACT(("firmware_cache released\n"));
+	dev_dbg(chip->card->dev, "firmware_cache released\n");
 #endif
 }
 
@@ -287,7 +290,7 @@ static int pcm_open(struct snd_pcm_substream *substream,
 
 	/* Set up hw capabilities and contraints */
 	memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware));
-	DE_HWP(("max_channels=%d\n", max_channels));
+	dev_dbg(chip->card->dev, "max_channels=%d\n", max_channels);
 	pipe->constr.list = channels_list;
 	pipe->constr.mask = 0;
 	for (i = 0; channels_list[i] <= max_channels; i++);
@@ -337,7 +340,7 @@ static int pcm_open(struct snd_pcm_substream *substream,
 	if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
 				       snd_dma_pci_data(chip->pci),
 				       PAGE_SIZE, &pipe->sgpage)) < 0) {
-		DE_HWP(("s-g list allocation failed\n"));
+		dev_err(chip->card->dev, "s-g list allocation failed\n");
 		return err;
 	}
 
@@ -351,7 +354,7 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream)
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
 	int err;
 
-	DE_ACT(("pcm_analog_in_open\n"));
+	dev_dbg(chip->card->dev, "pcm_analog_in_open\n");
 	if ((err = pcm_open(substream, num_analog_busses_in(chip) -
 			    substream->number)) < 0)
 		return err;
@@ -368,9 +371,9 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream)
 	atomic_inc(&chip->opencount);
 	if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
 		chip->can_set_rate=0;
-	DE_HWP(("pcm_analog_in_open  cs=%d  oc=%d  r=%d\n",
+	dev_dbg(chip->card->dev, "pcm_analog_in_open  cs=%d  oc=%d  r=%d\n",
 		chip->can_set_rate, atomic_read(&chip->opencount),
-		chip->sample_rate));
+		chip->sample_rate);
 	return 0;
 }
 
@@ -386,7 +389,7 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream)
 #else
 	max_channels = num_analog_busses_out(chip);
 #endif
-	DE_ACT(("pcm_analog_out_open\n"));
+	dev_dbg(chip->card->dev, "pcm_analog_out_open\n");
 	if ((err = pcm_open(substream, max_channels - substream->number)) < 0)
 		return err;
 	if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
@@ -404,9 +407,9 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream)
 	atomic_inc(&chip->opencount);
 	if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
 		chip->can_set_rate=0;
-	DE_HWP(("pcm_analog_out_open  cs=%d  oc=%d  r=%d\n",
+	dev_dbg(chip->card->dev, "pcm_analog_out_open  cs=%d  oc=%d  r=%d\n",
 		chip->can_set_rate, atomic_read(&chip->opencount),
-		chip->sample_rate));
+		chip->sample_rate);
 	return 0;
 }
 
@@ -419,7 +422,7 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream)
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
 	int err, max_channels;
 
-	DE_ACT(("pcm_digital_in_open\n"));
+	dev_dbg(chip->card->dev, "pcm_digital_in_open\n");
 	max_channels = num_digital_busses_in(chip) - substream->number;
 	mutex_lock(&chip->mode_mutex);
 	if (chip->digital_mode == DIGITAL_MODE_ADAT)
@@ -461,7 +464,7 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream)
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
 	int err, max_channels;
 
-	DE_ACT(("pcm_digital_out_open\n"));
+	dev_dbg(chip->card->dev, "pcm_digital_out_open\n");
 	max_channels = num_digital_busses_out(chip) - substream->number;
 	mutex_lock(&chip->mode_mutex);
 	if (chip->digital_mode == DIGITAL_MODE_ADAT)
@@ -508,18 +511,18 @@ static int pcm_close(struct snd_pcm_substream *substream)
 	/* Nothing to do here. Audio is already off and pipe will be
 	 * freed by its callback
 	 */
-	DE_ACT(("pcm_close\n"));
+	dev_dbg(chip->card->dev, "pcm_close\n");
 
 	atomic_dec(&chip->opencount);
 	oc = atomic_read(&chip->opencount);
-	DE_ACT(("pcm_close  oc=%d  cs=%d  rs=%d\n", oc,
-		chip->can_set_rate, chip->rate_set));
+	dev_dbg(chip->card->dev, "pcm_close  oc=%d  cs=%d  rs=%d\n", oc,
+		chip->can_set_rate, chip->rate_set);
 	if (oc < 2)
 		chip->can_set_rate = 1;
 	if (oc == 0)
 		chip->rate_set = 0;
-	DE_ACT(("pcm_close2 oc=%d  cs=%d  rs=%d\n", oc,
-		chip->can_set_rate,chip->rate_set));
+	dev_dbg(chip->card->dev, "pcm_close2 oc=%d  cs=%d  rs=%d\n", oc,
+		chip->can_set_rate, chip->rate_set);
 
 	return 0;
 }
@@ -543,7 +546,7 @@ static int init_engine(struct snd_pcm_substream *substream,
 	 */
 	spin_lock_irq(&chip->lock);
 	if (pipe->index >= 0) {
-		DE_HWP(("hwp_ie free(%d)\n", pipe->index));
+		dev_dbg(chip->card->dev, "hwp_ie free(%d)\n", pipe->index);
 		err = free_pipes(chip, pipe);
 		snd_BUG_ON(err);
 		chip->substream[pipe->index] = NULL;
@@ -552,16 +555,17 @@ static int init_engine(struct snd_pcm_substream *substream,
 	err = allocate_pipes(chip, pipe, pipe_index, interleave);
 	if (err < 0) {
 		spin_unlock_irq(&chip->lock);
-		DE_ACT((KERN_NOTICE "allocate_pipes(%d) err=%d\n",
-			pipe_index, err));
+		dev_err(chip->card->dev, "allocate_pipes(%d) err=%d\n",
+			pipe_index, err);
 		return err;
 	}
 	spin_unlock_irq(&chip->lock);
-	DE_ACT((KERN_NOTICE "allocate_pipes()=%d\n", pipe_index));
+	dev_dbg(chip->card->dev, "allocate_pipes()=%d\n", pipe_index);
 
-	DE_HWP(("pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n",
+	dev_dbg(chip->card->dev,
+		"pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n",
 		params_buffer_bytes(hw_params), params_periods(hw_params),
-		params_period_bytes(hw_params)));
+		params_period_bytes(hw_params));
 	err = snd_pcm_lib_malloc_pages(substream,
 				       params_buffer_bytes(hw_params));
 	if (err < 0) {
@@ -616,7 +620,7 @@ static int init_engine(struct snd_pcm_substream *substream,
 	spin_lock_irq(&chip->lock);
 	set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den);
 	spin_unlock_irq(&chip->lock);
-	DE_HWP(("pcm_hw_params ok\n"));
+	dev_dbg(chip->card->dev, "pcm_hw_params ok\n");
 	return 0;
 }
 
@@ -680,14 +684,14 @@ static int pcm_hw_free(struct snd_pcm_substream *substream)
 
 	spin_lock_irq(&chip->lock);
 	if (pipe->index >= 0) {
-		DE_HWP(("pcm_hw_free(%d)\n", pipe->index));
+		dev_dbg(chip->card->dev, "pcm_hw_free(%d)\n", pipe->index);
 		free_pipes(chip, pipe);
 		chip->substream[pipe->index] = NULL;
 		pipe->index = -1;
 	}
 	spin_unlock_irq(&chip->lock);
 
-	DE_HWP(("pcm_hw_freed\n"));
+	dev_dbg(chip->card->dev, "pcm_hw_freed\n");
 	snd_pcm_lib_free_pages(substream);
 	return 0;
 }
@@ -701,8 +705,8 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
 	struct audioformat format;
 	int pipe_index = ((struct audiopipe *)runtime->private_data)->index;
 
-	DE_HWP(("Prepare rate=%d format=%d channels=%d\n",
-		runtime->rate, runtime->format, runtime->channels));
+	dev_dbg(chip->card->dev, "Prepare rate=%d format=%d channels=%d\n",
+		runtime->rate, runtime->format, runtime->channels);
 	format.interleave = runtime->channels;
 	format.data_are_bigendian = 0;
 	format.mono_to_stereo = 0;
@@ -722,8 +726,9 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
 		format.bits_per_sample = 32;
 		break;
 	default:
-		DE_HWP(("Prepare error: unsupported format %d\n",
-			runtime->format));
+		dev_err(chip->card->dev,
+			"Prepare error: unsupported format %d\n",
+			runtime->format);
 		return -EINVAL;
 	}
 
@@ -758,10 +763,10 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	spin_lock(&chip->lock);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
-		DE_ACT(("pcm_trigger resume\n"));
+		dev_dbg(chip->card->dev, "pcm_trigger resume\n");
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		DE_ACT(("pcm_trigger start\n"));
+		dev_dbg(chip->card->dev, "pcm_trigger start\n");
 		for (i = 0; i < DSP_MAXPIPES; i++) {
 			if (channelmask & (1 << i)) {
 				pipe = chip->substream[i]->runtime->private_data;
@@ -783,9 +788,9 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 				      chip->pipe_cyclic_mask);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		DE_ACT(("pcm_trigger suspend\n"));
+		dev_dbg(chip->card->dev, "pcm_trigger suspend\n");
 	case SNDRV_PCM_TRIGGER_STOP:
-		DE_ACT(("pcm_trigger stop\n"));
+		dev_dbg(chip->card->dev, "pcm_trigger stop\n");
 		for (i = 0; i < DSP_MAXPIPES; i++) {
 			if (channelmask & (1 << i)) {
 				pipe = chip->substream[i]->runtime->private_data;
@@ -795,7 +800,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		err = stop_transport(chip, channelmask);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		DE_ACT(("pcm_trigger pause\n"));
+		dev_dbg(chip->card->dev, "pcm_trigger pause\n");
 		for (i = 0; i < DSP_MAXPIPES; i++) {
 			if (channelmask & (1 << i)) {
 				pipe = chip->substream[i]->runtime->private_data;
@@ -932,7 +937,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	DE_INIT(("Analog PCM ok\n"));
+	dev_dbg(chip->card->dev, "Analog PCM ok\n");
 
 #ifdef ECHOCARD_HAS_DIGITAL_IO
 	/* PCM#1 Digital inputs, no outputs */
@@ -945,7 +950,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	DE_INIT(("Digital PCM ok\n"));
+	dev_dbg(chip->card->dev, "Digital PCM ok\n");
 #endif /* ECHOCARD_HAS_DIGITAL_IO */
 
 #else /* ECHOCARD_HAS_VMIXER */
@@ -967,7 +972,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	DE_INIT(("Analog PCM ok\n"));
+	dev_dbg(chip->card->dev, "Analog PCM ok\n");
 
 #ifdef ECHOCARD_HAS_DIGITAL_IO
 	/* PCM#1 Digital i/o */
@@ -982,7 +987,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	DE_INIT(("Digital PCM ok\n"));
+	dev_dbg(chip->card->dev, "Digital PCM ok\n");
 #endif /* ECHOCARD_HAS_DIGITAL_IO */
 
 #endif /* ECHOCARD_HAS_VMIXER */
@@ -1475,7 +1480,8 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
 				snd_ctl_notify(chip->card,
 					       SNDRV_CTL_EVENT_MASK_VALUE,
 					       &chip->clock_src_ctl->id);
-				DE_ACT(("SDM() =%d\n", changed));
+				dev_dbg(chip->card->dev,
+					"SDM() =%d\n", changed);
 			}
 			if (changed >= 0)
 				changed = 1;	/* No errors */
@@ -1602,7 +1608,8 @@ static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol,
 	}
 
 	if (changed < 0)
-		DE_ACT(("seticlk val%d err 0x%x\n", dclock, changed));
+		dev_dbg(chip->card->dev,
+			"seticlk val%d err 0x%x\n", dclock, changed);
 
 	return changed;
 }
@@ -1859,7 +1866,7 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 #ifdef ECHOCARD_HAS_MIDI
 	if (st > 0 && chip->midi_in) {
 		snd_rawmidi_receive(chip->midi_in, chip->midi_buffer, st);
-		DE_MID(("rawmidi_iread=%d\n", st));
+		dev_dbg(chip->card->dev, "rawmidi_iread=%d\n", st);
 	}
 #endif
 	return IRQ_HANDLED;
@@ -1874,10 +1881,10 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 
 static int snd_echo_free(struct echoaudio *chip)
 {
-	DE_INIT(("Stop DSP...\n"));
+	dev_dbg(chip->card->dev, "Stop DSP...\n");
 	if (chip->comm_page)
 		rest_in_peace(chip);
-	DE_INIT(("Stopped.\n"));
+	dev_dbg(chip->card->dev, "Stopped.\n");
 
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1891,14 +1898,14 @@ static int snd_echo_free(struct echoaudio *chip)
 	if (chip->iores)
 		release_and_free_resource(chip->iores);
 
-	DE_INIT(("MMIO freed.\n"));
+	dev_dbg(chip->card->dev, "MMIO freed.\n");
 
 	pci_disable_device(chip->pci);
 
 	/* release chip data */
 	free_firmware_cache(chip);
 	kfree(chip);
-	DE_INIT(("Chip freed.\n"));
+	dev_dbg(chip->card->dev, "Chip freed.\n");
 	return 0;
 }
 
@@ -1908,7 +1915,7 @@ static int snd_echo_dev_free(struct snd_device *device)
 {
 	struct echoaudio *chip = device->device_data;
 
-	DE_INIT(("snd_echo_dev_free()...\n"));
+	dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n");
 	return snd_echo_free(chip);
 }
 
@@ -1941,7 +1948,7 @@ static int snd_echo_create(struct snd_card *card,
 			pci_disable_device(pci);
 			return -ENOMEM;
 		}
-		DE_INIT(("chip=%p\n", chip));
+		dev_dbg(card->dev, "chip=%p\n", chip);
 		spin_lock_init(&chip->lock);
 		chip->card = card;
 		chip->pci = pci;
@@ -1978,8 +1985,8 @@ static int snd_echo_create(struct snd_card *card,
 		return -EBUSY;
 	}
 	chip->irq = pci->irq;
-	DE_INIT(("pci=%p irq=%d subdev=%04x Init hardware...\n",
-		 chip->pci, chip->irq, chip->pci->subsystem_device));
+	dev_dbg(card->dev, "pci=%p irq=%d subdev=%04x Init hardware...\n",
+		chip->pci, chip->irq, chip->pci->subsystem_device);
 
 	/* Create the DSP comm page - this is the area of memory used for most
 	of the communication with the DSP, which accesses it via bus mastering */
@@ -1997,11 +2004,11 @@ static int snd_echo_create(struct snd_card *card,
 	if (err >= 0)
 		err = set_mixer_defaults(chip);
 	if (err < 0) {
-		DE_INIT(("init_hw err=%d\n", err));
+		dev_err(card->dev, "init_hw err=%d\n", err);
 		snd_echo_free(chip);
 		return err;
 	}
-	DE_INIT(("Card init OK\n"));
+	dev_dbg(card->dev, "Card init OK\n");
 
 	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
 		snd_echo_free(chip);
@@ -2031,7 +2038,7 @@ static int snd_echo_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	DE_INIT(("Echoaudio driver starting...\n"));
+	dev_dbg(&pci->dev, "Echoaudio driver starting...\n");
 	i = 0;
 	err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
 			   0, &card);
@@ -2184,7 +2191,7 @@ static int snd_echo_suspend(struct device *dev)
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct echoaudio *chip = dev_get_drvdata(dev);
 
-	DE_INIT(("suspend start\n"));
+	dev_dbg(dev, "suspend start\n");
 	snd_pcm_suspend_all(chip->analog_pcm);
 	snd_pcm_suspend_all(chip->digital_pcm);
 
@@ -2211,7 +2218,7 @@ static int snd_echo_suspend(struct device *dev)
 	pci_save_state(pci);
 	pci_disable_device(pci);
 
-	DE_INIT(("suspend done\n"));
+	dev_dbg(dev, "suspend done\n");
 	return 0;
 }
 
@@ -2225,7 +2232,7 @@ static int snd_echo_resume(struct device *dev)
 	u32 pipe_alloc_mask;
 	int err;
 
-	DE_INIT(("resume start\n"));
+	dev_dbg(dev, "resume start\n");
 	pci_restore_state(pci);
 	commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL);
 	if (commpage_bak == NULL)
@@ -2236,11 +2243,11 @@ static int snd_echo_resume(struct device *dev)
 	err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
 	if (err < 0) {
 		kfree(commpage_bak);
-		DE_INIT(("resume init_hw err=%d\n", err));
+		dev_err(dev, "resume init_hw err=%d\n", err);
 		snd_echo_free(chip);
 		return err;
 	}
-	DE_INIT(("resume init OK\n"));
+	dev_dbg(dev, "resume init OK\n");
 
 	/* Temporarily set chip->pipe_alloc_mask=0 otherwise
 	 * restore_dsp_settings() fails.
@@ -2253,7 +2260,7 @@ static int snd_echo_resume(struct device *dev)
 		kfree(commpage_bak);
 		return err;
 	}
-	DE_INIT(("resume restore OK\n"));
+	dev_dbg(dev, "resume restore OK\n");
 
 	memcpy(&commpage->audio_format, &commpage_bak->audio_format,
 		sizeof(commpage->audio_format));
@@ -2270,7 +2277,7 @@ static int snd_echo_resume(struct device *dev)
 		return -EBUSY;
 	}
 	chip->irq = pci->irq;
-	DE_INIT(("resume irq=%d\n", chip->irq));
+	dev_dbg(dev, "resume irq=%d\n", chip->irq);
 
 #ifdef ECHOCARD_HAS_MIDI
 	if (chip->midi_input_enabled)
@@ -2279,7 +2286,7 @@ static int snd_echo_resume(struct device *dev)
 		snd_echo_midi_output_trigger(chip->midi_out, 1);
 #endif
 
-	DE_INIT(("resume done\n"));
+	dev_dbg(dev, "resume done\n");
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index a4f112aa78e2..32515227a692 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -295,34 +295,6 @@
 #define PIPE_STATE_PENDING	3	/* Pipe has pending start */
 
 
-/* Debug initialization */
-#ifdef CONFIG_SND_DEBUG
-#define DE_INIT(x) snd_printk x
-#else
-#define DE_INIT(x)
-#endif
-
-/* Debug hw_params callbacks */
-#ifdef CONFIG_SND_DEBUG
-#define DE_HWP(x) snd_printk x
-#else
-#define DE_HWP(x)
-#endif
-
-/* Debug normal activity (open, start, stop...) */
-#ifdef CONFIG_SND_DEBUG
-#define DE_ACT(x) snd_printk x
-#else
-#define DE_ACT(x)
-#endif
-
-/* Debug midi activity */
-#ifdef CONFIG_SND_DEBUG
-#define DE_MID(x) snd_printk x
-#else
-#define DE_MID(x)
-#endif
-
 
 struct audiopipe {
 	volatile u32 *dma_counter;	/* Commpage register that contains
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
index 658db44ef746..50a21690447d 100644
--- a/sound/pci/echoaudio/echoaudio_3g.c
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -51,7 +51,7 @@ static int check_asic_status(struct echoaudio *chip)
 	}
 
 	box_status = le32_to_cpu(chip->comm_page->ext_box_status);
-	DE_INIT(("box_status=%x\n", box_status));
+	dev_dbg(chip->card->dev, "box_status=%x\n", box_status);
 	if (box_status == E3G_ASIC_NOT_LOADED)
 		return -ENODEV;
 
@@ -76,7 +76,8 @@ static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
 	if (wait_handshake(chip))
 		return -EIO;
 
-	DE_ACT(("WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq));
+	dev_dbg(chip->card->dev,
+		"WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq);
 
 	ctl = cpu_to_le32(ctl);
 	frq = cpu_to_le32(frq);
@@ -89,7 +90,7 @@ static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
 		return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
 	}
 
-	DE_ACT(("WriteControlReg: not written, no change\n"));
+	dev_dbg(chip->card->dev, "WriteControlReg: not written, no change\n");
 	return 0;
 }
 
@@ -258,8 +259,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	/* Only set the clock for internal mode. */
 	if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
-		DE_ACT(("set_sample_rate: Cannot set sample rate - "
-			"clock not set to CLK_CLOCKININTERNAL\n"));
+		dev_warn(chip->card->dev,
+			 "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n");
 		/* Save the rate anyhow */
 		chip->comm_page->sample_rate = cpu_to_le32(rate);
 		chip->sample_rate = rate;
@@ -313,7 +314,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	chip->comm_page->sample_rate = cpu_to_le32(rate);	/* ignored by the DSP */
 	chip->sample_rate = rate;
-	DE_ACT(("SetSampleRate: %d clock %x\n", rate, control_reg));
+	dev_dbg(chip->card->dev,
+		"SetSampleRate: %d clock %x\n", rate, control_reg);
 
 	/* Tell the DSP about it - DSP reads both control reg & freq reg */
 	return write_control_reg(chip, control_reg, frq_reg, 0);
@@ -326,7 +328,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
 	u32 control_reg, clocks_from_dsp;
 
-	DE_ACT(("set_input_clock:\n"));
+	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	/* Mask off the clock select bits */
 	control_reg = le32_to_cpu(chip->comm_page->control_register) &
@@ -335,13 +337,13 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		DE_ACT(("Set Echo3G clock to INTERNAL\n"));
+		dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
 		if (chip->digital_mode == DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		DE_ACT(("Set Echo3G clock to SPDIF\n"));
+		dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n");
 		control_reg |= E3G_SPDIF_CLOCK;
 		if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96)
 			control_reg |= E3G_DOUBLE_SPEED_MODE;
@@ -351,12 +353,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	case ECHO_CLOCK_ADAT:
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		DE_ACT(("Set Echo3G clock to ADAT\n"));
+		dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n");
 		control_reg |= E3G_ADAT_CLOCK;
 		control_reg &= ~E3G_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_WORD:
-		DE_ACT(("Set Echo3G clock to WORD\n"));
+		dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n");
 		control_reg |= E3G_WORD_CLOCK;
 		if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96)
 			control_reg |= E3G_DOUBLE_SPEED_MODE;
@@ -364,7 +366,8 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 			control_reg &= ~E3G_DOUBLE_SPEED_MODE;
 		break;
 	default:
-		DE_ACT(("Input clock 0x%x not supported for Echo3G\n", clock));
+		dev_err(chip->card->dev,
+			"Input clock 0x%x not supported for Echo3G\n", clock);
 		return -EINVAL;
 	}
 
@@ -392,7 +395,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 			incompatible_clock = TRUE;
 		break;
 	default:
-		DE_ACT(("Digital mode not supported: %d\n", mode));
+		dev_err(chip->card->dev,
+			"Digital mode not supported: %d\n", mode);
 		return -EINVAL;
 	}
 
@@ -427,6 +431,6 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 		return err;
 	chip->digital_mode = mode;
 
-	DE_ACT(("set_digital_mode(%d)\n", chip->digital_mode));
+	dev_dbg(chip->card->dev, "set_digital_mode(%d)\n", chip->digital_mode);
 	return incompatible_clock;
 }
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 977b2bd2e72f..ba9d4f16cbb3 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -80,7 +80,7 @@ static int send_vector(struct echoaudio *chip, u32 command)
 		udelay(1);
 	}
 
-	DE_ACT((KERN_ERR "timeout on send_vector\n"));
+	dev_err(chip->card->dev, "timeout on send_vector\n");
 	return -EBUSY;
 }
 
@@ -104,7 +104,7 @@ static int write_dsp(struct echoaudio *chip, u32 data)
 	}
 
 	chip->bad_board = TRUE;		/* Set TRUE until DSP re-loaded */
-	DE_ACT((KERN_ERR "write_dsp: Set bad_board to TRUE\n"));
+	dev_dbg(chip->card->dev, "write_dsp: Set bad_board to TRUE\n");
 	return -EIO;
 }
 
@@ -127,7 +127,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data)
 	}
 
 	chip->bad_board = TRUE;		/* Set TRUE until DSP re-loaded */
-	DE_INIT((KERN_ERR "read_dsp: Set bad_board to TRUE\n"));
+	dev_err(chip->card->dev, "read_dsp: Set bad_board to TRUE\n");
 	return -EIO;
 }
 
@@ -154,8 +154,9 @@ static int read_sn(struct echoaudio *chip)
 			return -EIO;
 		}
 	}
-	DE_INIT(("Read serial number %08x %08x %08x %08x %08x\n",
-		 sn[0], sn[1], sn[2], sn[3], sn[4]));
+	dev_dbg(chip->card->dev,
+		"Read serial number %08x %08x %08x %08x %08x\n",
+		 sn[0], sn[1], sn[2], sn[3], sn[4]);
 	return 0;
 }
 
@@ -205,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic)
 			goto la_error;
 	}
 
-	DE_INIT(("ASIC loaded\n"));
+	dev_dbg(chip->card->dev, "ASIC loaded\n");
 	free_firmware(fw, chip);
 	return 0;
 
 la_error:
-	DE_INIT(("failed on write_dsp\n"));
+	dev_err(chip->card->dev, "failed on write_dsp\n");
 	free_firmware(fw, chip);
 	return -EIO;
 }
@@ -241,8 +242,9 @@ static int install_resident_loader(struct echoaudio *chip)
 	loader is already installed, host flag 5 will be on. */
 	status = get_dsp_register(chip, CHI32_STATUS_REG);
 	if (status & CHI32_STATUS_REG_HF5) {
-		DE_INIT(("Resident loader already installed; status is 0x%x\n",
-			 status));
+		dev_dbg(chip->card->dev,
+			"Resident loader already installed; status is 0x%x\n",
+			 status);
 		return 0;
 	}
 
@@ -283,12 +285,14 @@ static int install_resident_loader(struct echoaudio *chip)
 
 	/* Write the count to the DSP */
 	if (write_dsp(chip, words)) {
-		DE_INIT(("install_resident_loader: Failed to write word count!\n"));
+		dev_err(chip->card->dev,
+			"install_resident_loader: Failed to write word count!\n");
 		goto irl_error;
 	}
 	/* Write the DSP address */
 	if (write_dsp(chip, address)) {
-		DE_INIT(("install_resident_loader: Failed to write DSP address!\n"));
+		dev_err(chip->card->dev,
+			"install_resident_loader: Failed to write DSP address!\n");
 		goto irl_error;
 	}
 	/* Write out this block of code to the DSP */
@@ -297,7 +301,8 @@ static int install_resident_loader(struct echoaudio *chip)
 
 		data = ((u32)code[index] << 16) + code[index + 1];
 		if (write_dsp(chip, data)) {
-			DE_INIT(("install_resident_loader: Failed to write DSP code\n"));
+			dev_err(chip->card->dev,
+				"install_resident_loader: Failed to write DSP code\n");
 			goto irl_error;
 		}
 		index += 2;
@@ -312,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip)
 	}
 
 	if (i == 200) {
-		DE_INIT(("Resident loader failed to set HF5\n"));
+		dev_err(chip->card->dev, "Resident loader failed to set HF5\n");
 		goto irl_error;
 	}
 
-	DE_INIT(("Resident loader successfully installed\n"));
+	dev_dbg(chip->card->dev, "Resident loader successfully installed\n");
 	free_firmware(fw, chip);
 	return 0;
 
@@ -334,14 +339,14 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 	int index, words, i;
 
 	if (chip->dsp_code == code) {
-		DE_INIT(("DSP is already loaded!\n"));
+		dev_warn(chip->card->dev, "DSP is already loaded!\n");
 		return 0;
 	}
 	chip->bad_board = TRUE;		/* Set TRUE until DSP loaded */
 	chip->dsp_code = NULL;		/* Current DSP code not loaded */
 	chip->asic_loaded = FALSE;	/* Loading the DSP code will reset the ASIC */
 
-	DE_INIT(("load_dsp: Set bad_board to TRUE\n"));
+	dev_dbg(chip->card->dev, "load_dsp: Set bad_board to TRUE\n");
 
 	/* If this board requires a resident loader, install it. */
 #ifdef DSP_56361
@@ -351,7 +356,8 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 
 	/* Send software reset command */
 	if (send_vector(chip, DSP_VC_RESET) < 0) {
-		DE_INIT(("LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n"));
+		dev_err(chip->card->dev,
+			"LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n");
 		return -EIO;
 	}
 	/* Delay 10us */
@@ -366,7 +372,8 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 	}
 
 	if (i == 1000) {
-		DE_INIT(("load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n"));
+		dev_err(chip->card->dev,
+			"load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n");
 		return -EIO;
 	}
 
@@ -403,29 +410,34 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 		index += 2;
 
 		if (write_dsp(chip, words) < 0) {
-			DE_INIT(("load_dsp: failed to write number of DSP words\n"));
+			dev_err(chip->card->dev,
+				"load_dsp: failed to write number of DSP words\n");
 			return -EIO;
 		}
 		if (write_dsp(chip, address) < 0) {
-			DE_INIT(("load_dsp: failed to write DSP address\n"));
+			dev_err(chip->card->dev,
+				"load_dsp: failed to write DSP address\n");
 			return -EIO;
 		}
 		if (write_dsp(chip, mem_type) < 0) {
-			DE_INIT(("load_dsp: failed to write DSP memory type\n"));
+			dev_err(chip->card->dev,
+				"load_dsp: failed to write DSP memory type\n");
 			return -EIO;
 		}
 		/* Code */
 		for (i = 0; i < words; i++, index+=2) {
 			data = ((u32)code[index] << 16) + code[index + 1];
 			if (write_dsp(chip, data) < 0) {
-				DE_INIT(("load_dsp: failed to write DSP data\n"));
+				dev_err(chip->card->dev,
+					"load_dsp: failed to write DSP data\n");
 				return -EIO;
 			}
 		}
 	}
 
 	if (write_dsp(chip, 0) < 0) {	/* We're done!!! */
-		DE_INIT(("load_dsp: Failed to write final zero\n"));
+		dev_err(chip->card->dev,
+			"load_dsp: Failed to write final zero\n");
 		return -EIO;
 	}
 	udelay(10);
@@ -438,12 +450,14 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 					 get_dsp_register(chip, CHI32_CONTROL_REG) & ~0x1b00);
 
 			if (write_dsp(chip, DSP_FNC_SET_COMMPAGE_ADDR) < 0) {
-				DE_INIT(("load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n"));
+				dev_err(chip->card->dev,
+					"load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n");
 				return -EIO;
 			}
 
 			if (write_dsp(chip, chip->comm_page_phys) < 0) {
-				DE_INIT(("load_dsp: Failed to write comm page address\n"));
+				dev_err(chip->card->dev,
+					"load_dsp: Failed to write comm page address\n");
 				return -EIO;
 			}
 
@@ -452,19 +466,21 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 			We don't actually use the serial number but we have to
 			get it as part of the DSP init voodoo. */
 			if (read_sn(chip) < 0) {
-				DE_INIT(("load_dsp: Failed to read serial number\n"));
+				dev_err(chip->card->dev,
+					"load_dsp: Failed to read serial number\n");
 				return -EIO;
 			}
 
 			chip->dsp_code = code;		/* Show which DSP code loaded */
 			chip->bad_board = FALSE;	/* DSP OK */
-			DE_INIT(("load_dsp: OK!\n"));
+			dev_dbg(chip->card->dev, "load_dsp: OK!\n");
 			return 0;
 		}
 		udelay(100);
 	}
 
-	DE_INIT(("load_dsp: DSP load timed out waiting for HF4\n"));
+	dev_err(chip->card->dev,
+		"load_dsp: DSP load timed out waiting for HF4\n");
 	return -EIO;
 }
 
@@ -658,7 +674,7 @@ static void get_audio_meters(struct echoaudio *chip, long *meters)
 static int restore_dsp_rettings(struct echoaudio *chip)
 {
 	int i, o, err;
-	DE_INIT(("restore_dsp_settings\n"));
+	dev_dbg(chip->card->dev, "restore_dsp_settings\n");
 
 	if ((err = check_asic_status(chip)) < 0)
 		return err;
@@ -755,7 +771,7 @@ static int restore_dsp_rettings(struct echoaudio *chip)
 	if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0)
 		return -EIO;
 
-	DE_INIT(("restore_dsp_rettings done\n"));
+	dev_dbg(chip->card->dev, "restore_dsp_rettings done\n");
 	return 0;
 }
 
@@ -835,7 +851,8 @@ static void set_audio_format(struct echoaudio *chip, u16 pipe_index,
 			break;
 		}
 	}
-	DE_ACT(("set_audio_format[%d] = %x\n", pipe_index, dsp_format));
+	dev_dbg(chip->card->dev,
+		 "set_audio_format[%d] = %x\n", pipe_index, dsp_format);
 	chip->comm_page->audio_format[pipe_index] = cpu_to_le16(dsp_format);
 }
 
@@ -848,7 +865,7 @@ Same thing for pause_ and stop_ -trasport below. */
 static int start_transport(struct echoaudio *chip, u32 channel_mask,
 			   u32 cyclic_mask)
 {
-	DE_ACT(("start_transport %x\n", channel_mask));
+	dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -866,7 +883,7 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask,
 		return 0;
 	}
 
-	DE_ACT(("start_transport: No pipes to start!\n"));
+	dev_err(chip->card->dev, "start_transport: No pipes to start!\n");
 	return -EINVAL;
 }
 
@@ -874,7 +891,7 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask,
 
 static int pause_transport(struct echoaudio *chip, u32 channel_mask)
 {
-	DE_ACT(("pause_transport %x\n", channel_mask));
+	dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -893,7 +910,7 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask)
 		return 0;
 	}
 
-	DE_ACT(("pause_transport: No pipes to stop!\n"));
+	dev_warn(chip->card->dev, "pause_transport: No pipes to stop!\n");
 	return 0;
 }
 
@@ -901,7 +918,7 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask)
 
 static int stop_transport(struct echoaudio *chip, u32 channel_mask)
 {
-	DE_ACT(("stop_transport %x\n", channel_mask));
+	dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -920,7 +937,7 @@ static int stop_transport(struct echoaudio *chip, u32 channel_mask)
 		return 0;
 	}
 
-	DE_ACT(("stop_transport: No pipes to stop!\n"));
+	dev_warn(chip->card->dev, "stop_transport: No pipes to stop!\n");
 	return 0;
 }
 
@@ -937,7 +954,8 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index)
 stopped and unallocated. */
 static int rest_in_peace(struct echoaudio *chip)
 {
-	DE_ACT(("rest_in_peace() open=%x\n", chip->pipe_alloc_mask));
+	dev_dbg(chip->card->dev,
+		"rest_in_peace() open=%x\n", chip->pipe_alloc_mask);
 
 	/* Stops all active pipes (just to be sure) */
 	stop_transport(chip, chip->active_mask);
@@ -965,7 +983,8 @@ static int init_dsp_comm_page(struct echoaudio *chip)
 {
 	/* Check if the compiler added extra padding inside the structure */
 	if (offsetof(struct comm_page, midi_output) != 0xbe0) {
-		DE_INIT(("init_dsp_comm_page() - Invalid struct comm_page structure\n"));
+		dev_err(chip->card->dev,
+			"init_dsp_comm_page() - Invalid struct comm_page structure\n");
 		return -EPERM;
 	}
 
@@ -999,7 +1018,7 @@ static int init_dsp_comm_page(struct echoaudio *chip)
  */
 static int init_line_levels(struct echoaudio *chip)
 {
-	DE_INIT(("init_line_levels\n"));
+	dev_dbg(chip->card->dev, "init_line_levels\n");
 	memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain));
 	memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain));
 	memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain));
@@ -1051,7 +1070,8 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
 	u32 channel_mask;
 	char is_cyclic;
 
-	DE_ACT(("allocate_pipes: ch=%d int=%d\n", pipe_index, interleave));
+	dev_dbg(chip->card->dev,
+		"allocate_pipes: ch=%d int=%d\n", pipe_index, interleave);
 
 	if (chip->bad_board)
 		return -EIO;
@@ -1061,7 +1081,8 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
 	for (channel_mask = i = 0; i < interleave; i++)
 		channel_mask |= 1 << (pipe_index + i);
 	if (chip->pipe_alloc_mask & channel_mask) {
-		DE_ACT(("allocate_pipes: channel already open\n"));
+		dev_err(chip->card->dev,
+			"allocate_pipes: channel already open\n");
 		return -EAGAIN;
 	}
 
@@ -1078,7 +1099,7 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
 	it moves data. The DMA counter is in units of bytes, not samples. */
 	pipe->dma_counter = &chip->comm_page->position[pipe_index];
 	*pipe->dma_counter = 0;
-	DE_ACT(("allocate_pipes: ok\n"));
+	dev_dbg(chip->card->dev, "allocate_pipes: ok\n");
 	return pipe_index;
 }
 
@@ -1089,7 +1110,7 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe)
 	u32 channel_mask;
 	int i;
 
-	DE_ACT(("free_pipes: Pipe %d\n", pipe->index));
+	dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index);
 	if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index)))
 		return -EINVAL;
 	if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED))
@@ -1131,7 +1152,7 @@ static int sglist_add_mapping(struct echoaudio *chip, struct audiopipe *pipe,
 		list[head].size = cpu_to_le32(length);
 		pipe->sglist_head++;
 	} else {
-		DE_ACT(("SGlist: too many fragments\n"));
+		dev_err(chip->card->dev, "SGlist: too many fragments\n");
 		return -ENOMEM;
 	}
 	return 0;
diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c
index afa273330e8a..23a099425834 100644
--- a/sound/pci/echoaudio/echoaudio_gml.c
+++ b/sound/pci/echoaudio/echoaudio_gml.c
@@ -46,7 +46,8 @@ static int check_asic_status(struct echoaudio *chip)
 	/* The DSP will return a value to indicate whether or not the
 	   ASIC is currently loaded */
 	if (read_dsp(chip, &asic_status) < 0) {
-		DE_INIT(("check_asic_status: failed on read_dsp\n"));
+		dev_err(chip->card->dev,
+			"check_asic_status: failed on read_dsp\n");
 		chip->asic_loaded = FALSE;
 		return -EIO;
 	}
@@ -68,7 +69,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force)
 	else
 		value &= ~GML_DIGITAL_IN_AUTO_MUTE;
 
-	DE_ACT(("write_control_reg: 0x%x\n", value));
+	dev_dbg(chip->card->dev, "write_control_reg: 0x%x\n", value);
 
 	/* Write the control register */
 	value = cpu_to_le32(value);
@@ -91,7 +92,7 @@ If the auto-mute is disabled, the digital inputs are enabled regardless of
 what the input clock is set or what is connected. */
 static int set_input_auto_mute(struct echoaudio *chip, int automute)
 {
-	DE_ACT(("set_input_auto_mute %d\n", automute));
+	dev_dbg(chip->card->dev, "set_input_auto_mute %d\n", automute);
 
 	chip->digital_in_automute = automute;
 
@@ -194,7 +195,7 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	if ((err = write_control_reg(chip, control_reg, FALSE)))
 		return err;
 	chip->professional_spdif = prof;
-	DE_ACT(("set_professional_spdif to %s\n",
-		prof ? "Professional" : "Consumer"));
+	dev_dbg(chip->card->dev, "set_professional_spdif to %s\n",
+		prof ? "Professional" : "Consumer");
 	return 0;
 }
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
index d1615a0579d1..a959eae95e0d 100644
--- a/sound/pci/echoaudio/gina20_dsp.c
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -37,12 +37,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Gina20\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Gina20\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -62,7 +63,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -149,7 +150,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
-	DE_ACT(("set_input_clock:\n"));
+	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
@@ -158,7 +159,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
 		set_sample_rate(chip, chip->sample_rate);
 		chip->input_clock = clock;
-		DE_ACT(("Set Gina clock to INTERNAL\n"));
+		dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n");
 		break;
 	case ECHO_CLOCK_SPDIF:
 		chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN;
@@ -166,7 +167,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		clear_handshake(chip);
 		send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
 		chip->clock_state = GD_CLOCK_SPDIFIN;
-		DE_ACT(("Set Gina20 clock to SPDIF\n"));
+		dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n");
 		chip->input_clock = clock;
 		break;
 	default:
@@ -208,7 +209,7 @@ static int update_flags(struct echoaudio *chip)
 
 static int set_professional_spdif(struct echoaudio *chip, char prof)
 {
-	DE_ACT(("set_professional_spdif %d\n", prof));
+	dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof);
 	if (prof)
 		chip->comm_page->flags |=
 			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c
index 98f7cfa81b5f..c8ea57612d22 100644
--- a/sound/pci/echoaudio/gina24_dsp.c
+++ b/sound/pci/echoaudio/gina24_dsp.c
@@ -41,12 +41,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Gina24\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Gina24\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -78,7 +79,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -155,7 +156,7 @@ static int load_asic(struct echoaudio *chip)
 		control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
 		err = write_control_reg(chip, control_reg, TRUE);
 	}
-	DE_INIT(("load_asic() done\n"));
+	dev_dbg(chip->card->dev, "load_asic() done\n");
 	return err;
 }
 
@@ -171,8 +172,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	/* Only set the clock for internal mode. */
 	if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
-		DE_ACT(("set_sample_rate: Cannot set sample rate - "
-			"clock not set to CLK_CLOCKININTERNAL\n"));
+		dev_warn(chip->card->dev,
+			 "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n");
 		/* Save the rate anyhow */
 		chip->comm_page->sample_rate = cpu_to_le32(rate);
 		chip->sample_rate = rate;
@@ -217,7 +218,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 		clock = GML_8KHZ;
 		break;
 	default:
-		DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+		dev_err(chip->card->dev,
+			"set_sample_rate: %d invalid!\n", rate);
 		return -EINVAL;
 	}
 
@@ -225,7 +227,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	chip->comm_page->sample_rate = cpu_to_le32(rate);	/* ignored by the DSP */
 	chip->sample_rate = rate;
-	DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+	dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, clock);
 
 	return write_control_reg(chip, control_reg, FALSE);
 }
@@ -236,7 +238,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
 	u32 control_reg, clocks_from_dsp;
 
-	DE_ACT(("set_input_clock:\n"));
+	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	/* Mask off the clock select bits */
 	control_reg = le32_to_cpu(chip->comm_page->control_register) &
@@ -245,13 +247,13 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		DE_ACT(("Set Gina24 clock to INTERNAL\n"));
+		dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
 		if (chip->digital_mode == DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		DE_ACT(("Set Gina24 clock to SPDIF\n"));
+		dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n");
 		control_reg |= GML_SPDIF_CLOCK;
 		if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
 			control_reg |= GML_DOUBLE_SPEED_MODE;
@@ -261,21 +263,22 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	case ECHO_CLOCK_ADAT:
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		DE_ACT(("Set Gina24 clock to ADAT\n"));
+		dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n");
 		control_reg |= GML_ADAT_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_ESYNC:
-		DE_ACT(("Set Gina24 clock to ESYNC\n"));
+		dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n");
 		control_reg |= GML_ESYNC_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_ESYNC96:
-		DE_ACT(("Set Gina24 clock to ESYNC96\n"));
+		dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n");
 		control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE;
 		break;
 	default:
-		DE_ACT(("Input clock 0x%x not supported for Gina24\n", clock));
+		dev_err(chip->card->dev,
+			"Input clock 0x%x not supported for Gina24\n", clock);
 		return -EINVAL;
 	}
 
@@ -304,7 +307,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 			incompatible_clock = TRUE;
 		break;
 	default:
-		DE_ACT(("Digital mode not supported: %d\n", mode));
+		dev_err(chip->card->dev,
+			"Digital mode not supported: %d\n", mode);
 		return -EINVAL;
 	}
 
@@ -344,6 +348,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 		return err;
 	chip->digital_mode = mode;
 
-	DE_ACT(("set_digital_mode to %d\n", chip->digital_mode));
+	dev_dbg(chip->card->dev,
+		"set_digital_mode to %d\n", chip->digital_mode);
 	return incompatible_clock;
 }
diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
index 5e85f14fe5a8..cdeb073fad85 100644
--- a/sound/pci/echoaudio/indigo_dsp.c
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -38,12 +38,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Indigo\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Indigo\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -60,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -109,7 +110,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 		control_reg = MIA_32000;
 		break;
 	default:
-		DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+		dev_err(chip->card->dev,
+			"set_sample_rate: %d invalid!\n", rate);
 		return -EINVAL;
 	}
 
@@ -147,7 +149,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
 	index = output * num_pipes_out(chip) + pipe;
 	chip->comm_page->vmixer[index] = gain;
 
-	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	dev_dbg(chip->card->dev,
+		"set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain);
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c
index 2e4ab3e34a74..ceda2d7046ac 100644
--- a/sound/pci/echoaudio/indigo_express_dsp.c
+++ b/sound/pci/echoaudio/indigo_express_dsp.c
@@ -61,7 +61,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	control_reg |= clock;
 	if (control_reg != old_control_reg) {
-		DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+		dev_dbg(chip->card->dev,
+			"set_sample_rate: %d clock %d\n", rate, clock);
 		chip->comm_page->control_register = cpu_to_le32(control_reg);
 		chip->sample_rate = rate;
 		clear_handshake(chip);
@@ -89,7 +90,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
 	index = output * num_pipes_out(chip) + pipe;
 	chip->comm_page->vmixer[index] = gain;
 
-	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	dev_dbg(chip->card->dev,
+		"set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain);
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
index 68f3c8ccc1bf..133915ca6438 100644
--- a/sound/pci/echoaudio/indigodj_dsp.c
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -38,12 +38,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Indigo DJ\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -60,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -109,7 +110,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 		control_reg = MIA_32000;
 		break;
 	default:
-		DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+		dev_err(chip->card->dev,
+			"set_sample_rate: %d invalid!\n", rate);
 		return -EINVAL;
 	}
 
@@ -147,7 +149,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
 	index = output * num_pipes_out(chip) + pipe;
 	chip->comm_page->vmixer[index] = gain;
 
-	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	dev_dbg(chip->card->dev,
+		"set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain);
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c
index bb9632c752a9..26cdfcfc5553 100644
--- a/sound/pci/echoaudio/indigodjx_dsp.c
+++ b/sound/pci/echoaudio/indigodjx_dsp.c
@@ -35,13 +35,14 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Indigo DJx\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX))
 		return -ENODEV;
 
 	err = init_dsp_comm_page(chip);
 	if (err < 0) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -59,7 +60,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
index beb9a5b69892..5e6df7c25055 100644
--- a/sound/pci/echoaudio/indigoio_dsp.c
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -38,12 +38,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Indigo IO\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -60,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -118,7 +119,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
 	index = output * num_pipes_out(chip) + pipe;
 	chip->comm_page->vmixer[index] = gain;
 
-	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	dev_dbg(chip->card->dev,
+		"set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain);
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c
index 394c6e76bcbc..90cdd271d9fc 100644
--- a/sound/pci/echoaudio/indigoiox_dsp.c
+++ b/sound/pci/echoaudio/indigoiox_dsp.c
@@ -35,13 +35,14 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Indigo IOx\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX))
 		return -ENODEV;
 
 	err = init_dsp_comm_page(chip);
 	if (err < 0) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -59,7 +60,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
index 53ce94605044..7f0f6ea08ca7 100644
--- a/sound/pci/echoaudio/layla20_dsp.c
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -40,12 +40,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Layla20\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Layla20\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -64,7 +65,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -121,7 +122,8 @@ static int check_asic_status(struct echoaudio *chip)
 		/* The DSP will return a value to indicate whether or not
 		   the ASIC is currently loaded */
 		if (read_dsp(chip, &asic_status) < 0) {
-			DE_ACT(("check_asic_status: failed on read_dsp\n"));
+			dev_err(chip->card->dev,
+				"check_asic_status: failed on read_dsp\n");
 			return -EIO;
 		}
 
@@ -164,8 +166,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 	/* Only set the clock for internal mode. Do not return failure,
 	   simply treat it as a non-event. */
 	if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
-		DE_ACT(("set_sample_rate: Cannot set sample rate - "
-			"clock not set to CLK_CLOCKININTERNAL\n"));
+		dev_warn(chip->card->dev,
+			 "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n");
 		chip->comm_page->sample_rate = cpu_to_le32(rate);
 		chip->sample_rate = rate;
 		return 0;
@@ -174,7 +176,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 	if (wait_handshake(chip))
 		return -EIO;
 
-	DE_ACT(("set_sample_rate(%d)\n", rate));
+	dev_dbg(chip->card->dev, "set_sample_rate(%d)\n", rate);
 	chip->sample_rate = rate;
 	chip->comm_page->sample_rate = cpu_to_le32(rate);
 	clear_handshake(chip);
@@ -188,29 +190,30 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source)
 	u16 clock;
 	u32 rate;
 
-	DE_ACT(("set_input_clock:\n"));
+	dev_dbg(chip->card->dev, "set_input_clock:\n");
 	rate = 0;
 	switch (clock_source) {
 	case ECHO_CLOCK_INTERNAL:
-		DE_ACT(("Set Layla20 clock to INTERNAL\n"));
+		dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n");
 		rate = chip->sample_rate;
 		clock = LAYLA20_CLOCK_INTERNAL;
 		break;
 	case ECHO_CLOCK_SPDIF:
-		DE_ACT(("Set Layla20 clock to SPDIF\n"));
+		dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n");
 		clock = LAYLA20_CLOCK_SPDIF;
 		break;
 	case ECHO_CLOCK_WORD:
-		DE_ACT(("Set Layla20 clock to WORD\n"));
+		dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n");
 		clock = LAYLA20_CLOCK_WORD;
 		break;
 	case ECHO_CLOCK_SUPER:
-		DE_ACT(("Set Layla20 clock to SUPER\n"));
+		dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n");
 		clock = LAYLA20_CLOCK_SUPER;
 		break;
 	default:
-		DE_ACT(("Input clock 0x%x not supported for Layla24\n",
-			clock_source));
+		dev_err(chip->card->dev,
+			"Input clock 0x%x not supported for Layla24\n",
+			clock_source);
 		return -EINVAL;
 	}
 	chip->input_clock = clock_source;
@@ -229,7 +232,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source)
 
 static int set_output_clock(struct echoaudio *chip, u16 clock)
 {
-	DE_ACT(("set_output_clock: %d\n", clock));
+	dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock);
 	switch (clock) {
 	case ECHO_CLOCK_SUPER:
 		clock = LAYLA20_OUTPUT_CLOCK_SUPER;
@@ -238,7 +241,7 @@ static int set_output_clock(struct echoaudio *chip, u16 clock)
 		clock = LAYLA20_OUTPUT_CLOCK_WORD;
 		break;
 	default:
-		DE_ACT(("set_output_clock wrong clock\n"));
+		dev_err(chip->card->dev, "set_output_clock wrong clock\n");
 		return -EINVAL;
 	}
 
@@ -283,7 +286,7 @@ static int update_flags(struct echoaudio *chip)
 
 static int set_professional_spdif(struct echoaudio *chip, char prof)
 {
-	DE_ACT(("set_professional_spdif %d\n", prof));
+	dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof);
 	if (prof)
 		chip->comm_page->flags |=
 			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c
index 8c041647f285..eb8f218f79fe 100644
--- a/sound/pci/echoaudio/layla24_dsp.c
+++ b/sound/pci/echoaudio/layla24_dsp.c
@@ -40,12 +40,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Layla24\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Layla24\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -69,7 +70,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -117,7 +118,7 @@ static int load_asic(struct echoaudio *chip)
 	if (chip->asic_loaded)
 		return 1;
 
-	DE_INIT(("load_asic\n"));
+	dev_dbg(chip->card->dev, "load_asic\n");
 
 	/* Give the DSP a few milliseconds to settle down */
 	mdelay(10);
@@ -151,7 +152,7 @@ static int load_asic(struct echoaudio *chip)
 		err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ,
 					TRUE);
 	
-	DE_INIT(("load_asic() done\n"));
+	dev_dbg(chip->card->dev, "load_asic() done\n");
 	return err;
 }
 
@@ -167,8 +168,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	/* Only set the clock for internal mode. */
 	if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
-		DE_ACT(("set_sample_rate: Cannot set sample rate - "
-			"clock not set to CLK_CLOCKININTERNAL\n"));
+		dev_warn(chip->card->dev,
+			 "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n");
 		/* Save the rate anyhow */
 		chip->comm_page->sample_rate = cpu_to_le32(rate);
 		chip->sample_rate = rate;
@@ -241,7 +242,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	chip->comm_page->sample_rate = cpu_to_le32(rate);	/* ignored by the DSP ? */
 	chip->sample_rate = rate;
-	DE_ACT(("set_sample_rate: %d clock %d\n", rate, control_reg));
+	dev_dbg(chip->card->dev,
+		"set_sample_rate: %d clock %d\n", rate, control_reg);
 
 	return write_control_reg(chip, control_reg, FALSE);
 }
@@ -260,7 +262,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	/* Pick the new clock */
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		DE_ACT(("Set Layla24 clock to INTERNAL\n"));
+		dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
@@ -269,7 +271,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		control_reg |= GML_SPDIF_CLOCK;
 		/* Layla24 doesn't support 96KHz S/PDIF */
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
-		DE_ACT(("Set Layla24 clock to SPDIF\n"));
+		dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n");
 		break;
 	case ECHO_CLOCK_WORD:
 		control_reg |= GML_WORD_CLOCK;
@@ -277,17 +279,18 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 			control_reg |= GML_DOUBLE_SPEED_MODE;
 		else
 			control_reg &= ~GML_DOUBLE_SPEED_MODE;
-		DE_ACT(("Set Layla24 clock to WORD\n"));
+		dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n");
 		break;
 	case ECHO_CLOCK_ADAT:
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
 		control_reg |= GML_ADAT_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
-		DE_ACT(("Set Layla24 clock to ADAT\n"));
+		dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n");
 		break;
 	default:
-		DE_ACT(("Input clock 0x%x not supported for Layla24\n", clock));
+		dev_err(chip->card->dev,
+			"Input clock 0x%x not supported for Layla24\n", clock);
 		return -EINVAL;
 	}
 
@@ -353,7 +356,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 		asic = FW_LAYLA24_2A_ASIC;
 		break;
 	default:
-		DE_ACT(("Digital mode not supported: %d\n", mode));
+		dev_err(chip->card->dev,
+			"Digital mode not supported: %d\n", mode);
 		return -EINVAL;
 	}
 
@@ -393,6 +397,6 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 		return err;
 	chip->digital_mode = mode;
 
-	DE_ACT(("set_digital_mode to %d\n", mode));
+	dev_dbg(chip->card->dev, "set_digital_mode to %d\n", mode);
 	return incompatible_clock;
 }
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
index 6ebfa6e7ab9e..ed2f21dcd1c9 100644
--- a/sound/pci/echoaudio/mia_dsp.c
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -41,12 +41,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Mia\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Mia\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -66,7 +67,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -126,7 +127,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 		control_reg = MIA_32000;
 		break;
 	default:
-		DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+		dev_err(chip->card->dev,
+			"set_sample_rate: %d invalid!\n", rate);
 		return -EINVAL;
 	}
 
@@ -153,7 +155,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
-	DE_ACT(("set_input_clock(%d)\n", clock));
+	dev_dbg(chip->card->dev, "set_input_clock(%d)\n", clock);
 	if (snd_BUG_ON(clock != ECHO_CLOCK_INTERNAL &&
 		       clock != ECHO_CLOCK_SPDIF))
 		return -EINVAL;
@@ -181,7 +183,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
 	index = output * num_pipes_out(chip) + pipe;
 	chip->comm_page->vmixer[index] = gain;
 
-	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	dev_dbg(chip->card->dev,
+		"set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain);
 	return 0;
 }
 
@@ -211,7 +214,7 @@ static int update_flags(struct echoaudio *chip)
 
 static int set_professional_spdif(struct echoaudio *chip, char prof)
 {
-	DE_ACT(("set_professional_spdif %d\n", prof));
+	dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof);
 	if (prof)
 		chip->comm_page->flags |=
 			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
index 7f4dfae0323a..8d43c5a4976b 100644
--- a/sound/pci/echoaudio/midi.c
+++ b/sound/pci/echoaudio/midi.c
@@ -36,7 +36,7 @@
 /* Start and stop Midi input */
 static int enable_midi_input(struct echoaudio *chip, char enable)
 {
-	DE_MID(("enable_midi_input(%d)\n", enable));
+	dev_dbg(chip->card->dev, "enable_midi_input(%d)\n", enable);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -74,7 +74,7 @@ static int write_midi(struct echoaudio *chip, u8 *data, int bytes)
 	chip->comm_page->midi_out_free_count = 0;
 	clear_handshake(chip);
 	send_vector(chip, DSP_VC_MIDI_WRITE);
-	DE_MID(("write_midi: %d\n", bytes));
+	dev_dbg(chip->card->dev, "write_midi: %d\n", bytes);
 	return bytes;
 }
 
@@ -157,7 +157,7 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream)
 	struct echoaudio *chip = substream->rmidi->private_data;
 
 	chip->midi_in = substream;
-	DE_MID(("rawmidi_iopen\n"));
+	dev_dbg(chip->card->dev, "rawmidi_iopen\n");
 	return 0;
 }
 
@@ -183,7 +183,7 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream)
 	struct echoaudio *chip = substream->rmidi->private_data;
 
 	chip->midi_in = NULL;
-	DE_MID(("rawmidi_iclose\n"));
+	dev_dbg(chip->card->dev, "rawmidi_iclose\n");
 	return 0;
 }
 
@@ -196,7 +196,7 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream)
 	chip->tinuse = 0;
 	chip->midi_full = 0;
 	chip->midi_out = substream;
-	DE_MID(("rawmidi_oopen\n"));
+	dev_dbg(chip->card->dev, "rawmidi_open\n");
 	return 0;
 }
 
@@ -209,7 +209,7 @@ static void snd_echo_midi_output_write(unsigned long data)
 	int bytes, sent, time;
 	unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1];
 
-	DE_MID(("snd_echo_midi_output_write\n"));
+	dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n");
 	/* No interrupts are involved: we have to check at regular intervals
 	if the card's output buffer has room for new data. */
 	sent = bytes = 0;
@@ -218,7 +218,7 @@ static void snd_echo_midi_output_write(unsigned long data)
 	if (!snd_rawmidi_transmit_empty(chip->midi_out)) {
 		bytes = snd_rawmidi_transmit_peek(chip->midi_out, buf,
 						  MIDI_OUT_BUFFER_SIZE - 1);
-		DE_MID(("Try to send %d bytes...\n", bytes));
+		dev_dbg(chip->card->dev, "Try to send %d bytes...\n", bytes);
 		sent = write_midi(chip, buf, bytes);
 		if (sent < 0) {
 			dev_err(chip->card->dev,
@@ -227,12 +227,12 @@ static void snd_echo_midi_output_write(unsigned long data)
 			sent = 9000;
 			chip->midi_full = 1;
 		} else if (sent > 0) {
-			DE_MID(("%d bytes sent\n", sent));
+			dev_dbg(chip->card->dev, "%d bytes sent\n", sent);
 			snd_rawmidi_transmit_ack(chip->midi_out, sent);
 		} else {
 			/* Buffer is full. DSP's internal buffer is 64 (128 ?)
 			bytes long. Let's wait until half of them are sent */
-			DE_MID(("Full\n"));
+			dev_dbg(chip->card->dev, "Full\n");
 			sent = 32;
 			chip->midi_full = 1;
 		}
@@ -244,7 +244,8 @@ static void snd_echo_midi_output_write(unsigned long data)
 		   sent */
 		time = (sent << 3) / 25 + 1;	/* 8/25=0.32ms to send a byte */
 		mod_timer(&chip->timer, jiffies + (time * HZ + 999) / 1000);
-		DE_MID(("Timer armed(%d)\n", ((time * HZ + 999) / 1000)));
+		dev_dbg(chip->card->dev,
+			"Timer armed(%d)\n", ((time * HZ + 999) / 1000));
 	}
 	spin_unlock_irqrestore(&chip->lock, flags);
 }
@@ -256,7 +257,7 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream
 {
 	struct echoaudio *chip = substream->rmidi->private_data;
 
-	DE_MID(("snd_echo_midi_output_trigger(%d)\n", up));
+	dev_dbg(chip->card->dev, "snd_echo_midi_output_trigger(%d)\n", up);
 	spin_lock_irq(&chip->lock);
 	if (up) {
 		if (!chip->tinuse) {
@@ -270,7 +271,7 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream
 			chip->tinuse = 0;
 			spin_unlock_irq(&chip->lock);
 			del_timer_sync(&chip->timer);
-			DE_MID(("Timer removed\n"));
+			dev_dbg(chip->card->dev, "Timer removed\n");
 			return;
 		}
 	}
@@ -287,7 +288,7 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream)
 	struct echoaudio *chip = substream->rmidi->private_data;
 
 	chip->midi_out = NULL;
-	DE_MID(("rawmidi_oclose\n"));
+	dev_dbg(chip->card->dev, "rawmidi_oclose\n");
 	return 0;
 }
 
@@ -327,6 +328,6 @@ static int snd_echo_midi_create(struct snd_card *card,
 
 	chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
 		SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
-	DE_INIT(("MIDI ok\n"));
+	dev_dbg(chip->card->dev, "MIDI ok\n");
 	return 0;
 }
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
index 6e6a7eb555b8..cc46a8c8e3dd 100644
--- a/sound/pci/echoaudio/mona_dsp.c
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -41,12 +41,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	DE_INIT(("init_hw() - Mona\n"));
+	dev_dbg(chip->card->dev, "init_hw() - Mona\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA))
 		return -ENODEV;
 
 	if ((err = init_dsp_comm_page(chip))) {
-		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		dev_err(chip->card->dev,
+			"init_hw - could not initialize DSP comm page\n");
 		return err;
 	}
 
@@ -71,7 +72,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	DE_INIT(("init_hw done\n"));
+	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -202,8 +203,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	/* Only set the clock for internal mode. */
 	if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
-		DE_ACT(("set_sample_rate: Cannot set sample rate - "
-			"clock not set to CLK_CLOCKININTERNAL\n"));
+		dev_dbg(chip->card->dev,
+			"Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n");
 		/* Save the rate anyhow */
 		chip->comm_page->sample_rate = cpu_to_le32(rate);
 		chip->sample_rate = rate;
@@ -279,7 +280,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 		clock = GML_8KHZ;
 		break;
 	default:
-		DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+		dev_err(chip->card->dev,
+			"set_sample_rate: %d invalid!\n", rate);
 		return -EINVAL;
 	}
 
@@ -287,7 +289,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 	chip->comm_page->sample_rate = cpu_to_le32(rate);	/* ignored by the DSP */
 	chip->sample_rate = rate;
-	DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+	dev_dbg(chip->card->dev,
+		"set_sample_rate: %d clock %d\n", rate, clock);
 
 	return write_control_reg(chip, control_reg, force_write);
 }
@@ -299,7 +302,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	u32 control_reg, clocks_from_dsp;
 	int err;
 
-	DE_ACT(("set_input_clock:\n"));
+	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	/* Prevent two simultaneous calls to switch_asic() */
 	if (atomic_read(&chip->opencount))
@@ -312,7 +315,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		DE_ACT(("Set Mona clock to INTERNAL\n"));
+		dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
@@ -324,7 +327,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		spin_lock_irq(&chip->lock);
 		if (err < 0)
 			return err;
-		DE_ACT(("Set Mona clock to SPDIF\n"));
+		dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n");
 		control_reg |= GML_SPDIF_CLOCK;
 		if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
 			control_reg |= GML_DOUBLE_SPEED_MODE;
@@ -332,7 +335,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 			control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_WORD:
-		DE_ACT(("Set Mona clock to WORD\n"));
+		dev_dbg(chip->card->dev, "Set Mona clock to WORD\n");
 		spin_unlock_irq(&chip->lock);
 		err = switch_asic(chip, clocks_from_dsp &
 				  GML_CLOCK_DETECT_BIT_WORD96);
@@ -346,14 +349,15 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 			control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_ADAT:
-		DE_ACT(("Set Mona clock to ADAT\n"));
+		dev_dbg(chip->card->dev, "Set Mona clock to ADAT\n");
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
 		control_reg |= GML_ADAT_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	default:
-		DE_ACT(("Input clock 0x%x not supported for Mona\n", clock));
+		dev_err(chip->card->dev,
+			"Input clock 0x%x not supported for Mona\n", clock);
 		return -EINVAL;
 	}
 
@@ -381,7 +385,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 			incompatible_clock = TRUE;
 		break;
 	default:
-		DE_ACT(("Digital mode not supported: %d\n", mode));
+		dev_err(chip->card->dev,
+			"Digital mode not supported: %d\n", mode);
 		return -EINVAL;
 	}
 
@@ -422,6 +427,6 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
 		return err;
 	chip->digital_mode = mode;
 
-	DE_ACT(("set_digital_mode to %d\n", mode));
+	dev_dbg(chip->card->dev, "set_digital_mode to %d\n", mode);
 	return incompatible_clock;
 }
-- 
cgit v1.2.3


From 31604d35db18c1382c7ee9fa836ff9ab0b4d2751 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Mon, 3 Nov 2014 14:54:36 +0100
Subject: ALSA: emu10k1: Deletion of unnecessary checks before three function
 calls

The functions kfree(), release_firmware() and snd_util_memhdr_free() test
whether their argument is NULL and then return immediately. Thus the test
around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/emu10k1_main.c | 9 +++------
 sound/pci/emu10k1/emufx.c        | 3 +--
 2 files changed, 4 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 229269788023..b4458a630a7c 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1289,10 +1289,8 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
 	}
 	if (emu->emu1010.firmware_thread)
 		kthread_stop(emu->emu1010.firmware_thread);
-	if (emu->firmware)
-		release_firmware(emu->firmware);
-	if (emu->dock_fw)
-		release_firmware(emu->dock_fw);
+	release_firmware(emu->firmware);
+	release_firmware(emu->dock_fw);
 	if (emu->irq >= 0)
 		free_irq(emu->irq, emu);
 	/* remove reserved page */
@@ -1301,8 +1299,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
 			(struct snd_util_memblk *)emu->reserved_page);
 		emu->reserved_page = NULL;
 	}
-	if (emu->memhdr)
-		snd_util_memhdr_free(emu->memhdr);
+	snd_util_memhdr_free(emu->memhdr);
 	if (emu->silent_page.area)
 		snd_dma_free_pages(&emu->silent_page);
 	if (emu->ptb_pages.area)
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 745f0627c634..eb5c0aba41c1 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -777,8 +777,7 @@ static void snd_emu10k1_ctl_private_free(struct snd_kcontrol *kctl)
 	kctl->private_value = 0;
 	list_del(&ctl->list);
 	kfree(ctl);
-	if (kctl->tlv.p)
-		kfree(kctl->tlv.p);
+	kfree(kctl->tlv.p);
 }
 
 static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu,
-- 
cgit v1.2.3


From 9161bd0d1cf375492f0a6aa86b3e4c28b070fb7c Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Date: Wed, 5 Nov 2014 19:51:56 +0530
Subject: ALSA: echoaudio: cleanup of unnecessary messages

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing
chip after it has been freed. This patch fixes that and at the same
time removes some debugging messages, which are unnecessary, as they
are just printing information about entry and exit from a function,
and which switch-case it is executing.
we can easily get from ftrace the information about the entry and exit
from a function.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/darla20_dsp.c   |  2 --
 sound/pci/echoaudio/darla24_dsp.c   |  2 --
 sound/pci/echoaudio/echo3g_dsp.c    |  2 --
 sound/pci/echoaudio/echoaudio.c     | 31 -------------------------------
 sound/pci/echoaudio/echoaudio_3g.c  |  5 -----
 sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------
 sound/pci/echoaudio/gina20_dsp.c    |  6 ------
 sound/pci/echoaudio/gina24_dsp.c    |  9 ---------
 sound/pci/echoaudio/indigo_dsp.c    |  2 --
 sound/pci/echoaudio/indigodj_dsp.c  |  2 --
 sound/pci/echoaudio/indigodjx_dsp.c |  2 --
 sound/pci/echoaudio/indigoio_dsp.c  |  2 --
 sound/pci/echoaudio/indigoiox_dsp.c |  2 --
 sound/pci/echoaudio/layla20_dsp.c   |  9 ---------
 sound/pci/echoaudio/layla24_dsp.c   |  8 --------
 sound/pci/echoaudio/mia_dsp.c       |  2 --
 sound/pci/echoaudio/midi.c          |  6 ------
 sound/pci/echoaudio/mona_dsp.c      |  6 ------
 18 files changed, 110 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c
index c94e92e31ae6..febee5bda877 100644
--- a/sound/pci/echoaudio/darla20_dsp.c
+++ b/sound/pci/echoaudio/darla20_dsp.c
@@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Darla20\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20))
 		return -ENODEV;
 
@@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw: done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c
index b1272f88d59d..7b4f6fd51b09 100644
--- a/sound/pci/echoaudio/darla24_dsp.c
+++ b/sound/pci/echoaudio/darla24_dsp.c
@@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Darla24\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24))
 		return -ENODEV;
 
@@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw: done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
index bc3716895fc8..ae11ce11b1c2 100644
--- a/sound/pci/echoaudio/echo3g_dsp.c
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	int err;
 
 	local_irq_enable();
-	dev_dbg(chip->card->dev, "init_hw() - Echo3G\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G))
 		return -ENODEV;
 
@@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 				ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
 				ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 1ef29e5b53a7..60e40034b991 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -79,7 +79,6 @@ static void free_firmware(const struct firmware *fw_entry,
 	dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n");
 #else
 	release_firmware(fw_entry);
-	dev_dbg(chip->card->dev, "firmware released\n");
 #endif
 }
 
@@ -96,7 +95,6 @@ static void free_firmware_cache(struct echoaudio *chip)
 			dev_dbg(chip->card->dev, "release_firmware(%d)\n", i);
 		}
 
-	dev_dbg(chip->card->dev, "firmware_cache released\n");
 #endif
 }
 
@@ -354,7 +352,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream)
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
 	int err;
 
-	dev_dbg(chip->card->dev, "pcm_analog_in_open\n");
 	if ((err = pcm_open(substream, num_analog_busses_in(chip) -
 			    substream->number)) < 0)
 		return err;
@@ -389,7 +386,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream)
 #else
 	max_channels = num_analog_busses_out(chip);
 #endif
-	dev_dbg(chip->card->dev, "pcm_analog_out_open\n");
 	if ((err = pcm_open(substream, max_channels - substream->number)) < 0)
 		return err;
 	if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
@@ -422,7 +418,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream)
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
 	int err, max_channels;
 
-	dev_dbg(chip->card->dev, "pcm_digital_in_open\n");
 	max_channels = num_digital_busses_in(chip) - substream->number;
 	mutex_lock(&chip->mode_mutex);
 	if (chip->digital_mode == DIGITAL_MODE_ADAT)
@@ -464,7 +459,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream)
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
 	int err, max_channels;
 
-	dev_dbg(chip->card->dev, "pcm_digital_out_open\n");
 	max_channels = num_digital_busses_out(chip) - substream->number;
 	mutex_lock(&chip->mode_mutex);
 	if (chip->digital_mode == DIGITAL_MODE_ADAT)
@@ -511,7 +505,6 @@ static int pcm_close(struct snd_pcm_substream *substream)
 	/* Nothing to do here. Audio is already off and pipe will be
 	 * freed by its callback
 	 */
-	dev_dbg(chip->card->dev, "pcm_close\n");
 
 	atomic_dec(&chip->opencount);
 	oc = atomic_read(&chip->opencount);
@@ -620,7 +613,6 @@ static int init_engine(struct snd_pcm_substream *substream,
 	spin_lock_irq(&chip->lock);
 	set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den);
 	spin_unlock_irq(&chip->lock);
-	dev_dbg(chip->card->dev, "pcm_hw_params ok\n");
 	return 0;
 }
 
@@ -691,7 +683,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream)
 	}
 	spin_unlock_irq(&chip->lock);
 
-	dev_dbg(chip->card->dev, "pcm_hw_freed\n");
 	snd_pcm_lib_free_pages(substream);
 	return 0;
 }
@@ -763,10 +754,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	spin_lock(&chip->lock);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
-		dev_dbg(chip->card->dev, "pcm_trigger resume\n");
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		dev_dbg(chip->card->dev, "pcm_trigger start\n");
 		for (i = 0; i < DSP_MAXPIPES; i++) {
 			if (channelmask & (1 << i)) {
 				pipe = chip->substream[i]->runtime->private_data;
@@ -788,9 +777,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 				      chip->pipe_cyclic_mask);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		dev_dbg(chip->card->dev, "pcm_trigger suspend\n");
 	case SNDRV_PCM_TRIGGER_STOP:
-		dev_dbg(chip->card->dev, "pcm_trigger stop\n");
 		for (i = 0; i < DSP_MAXPIPES; i++) {
 			if (channelmask & (1 << i)) {
 				pipe = chip->substream[i]->runtime->private_data;
@@ -800,7 +787,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		err = stop_transport(chip, channelmask);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		dev_dbg(chip->card->dev, "pcm_trigger pause\n");
 		for (i = 0; i < DSP_MAXPIPES; i++) {
 			if (channelmask & (1 << i)) {
 				pipe = chip->substream[i]->runtime->private_data;
@@ -937,7 +923,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	dev_dbg(chip->card->dev, "Analog PCM ok\n");
 
 #ifdef ECHOCARD_HAS_DIGITAL_IO
 	/* PCM#1 Digital inputs, no outputs */
@@ -950,7 +935,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	dev_dbg(chip->card->dev, "Digital PCM ok\n");
 #endif /* ECHOCARD_HAS_DIGITAL_IO */
 
 #else /* ECHOCARD_HAS_VMIXER */
@@ -972,7 +956,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	dev_dbg(chip->card->dev, "Analog PCM ok\n");
 
 #ifdef ECHOCARD_HAS_DIGITAL_IO
 	/* PCM#1 Digital i/o */
@@ -987,7 +970,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
 	if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
 		return err;
-	dev_dbg(chip->card->dev, "Digital PCM ok\n");
 #endif /* ECHOCARD_HAS_DIGITAL_IO */
 
 #endif /* ECHOCARD_HAS_VMIXER */
@@ -1881,10 +1863,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 
 static int snd_echo_free(struct echoaudio *chip)
 {
-	dev_dbg(chip->card->dev, "Stop DSP...\n");
 	if (chip->comm_page)
 		rest_in_peace(chip);
-	dev_dbg(chip->card->dev, "Stopped.\n");
 
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1898,14 +1878,12 @@ static int snd_echo_free(struct echoaudio *chip)
 	if (chip->iores)
 		release_and_free_resource(chip->iores);
 
-	dev_dbg(chip->card->dev, "MMIO freed.\n");
 
 	pci_disable_device(chip->pci);
 
 	/* release chip data */
 	free_firmware_cache(chip);
 	kfree(chip);
-	dev_dbg(chip->card->dev, "Chip freed.\n");
 	return 0;
 }
 
@@ -1915,7 +1893,6 @@ static int snd_echo_dev_free(struct snd_device *device)
 {
 	struct echoaudio *chip = device->device_data;
 
-	dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n");
 	return snd_echo_free(chip);
 }
 
@@ -2008,7 +1985,6 @@ static int snd_echo_create(struct snd_card *card,
 		snd_echo_free(chip);
 		return err;
 	}
-	dev_dbg(card->dev, "Card init OK\n");
 
 	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
 		snd_echo_free(chip);
@@ -2038,7 +2014,6 @@ static int snd_echo_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	dev_dbg(&pci->dev, "Echoaudio driver starting...\n");
 	i = 0;
 	err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
 			   0, &card);
@@ -2191,7 +2166,6 @@ static int snd_echo_suspend(struct device *dev)
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct echoaudio *chip = dev_get_drvdata(dev);
 
-	dev_dbg(dev, "suspend start\n");
 	snd_pcm_suspend_all(chip->analog_pcm);
 	snd_pcm_suspend_all(chip->digital_pcm);
 
@@ -2218,7 +2192,6 @@ static int snd_echo_suspend(struct device *dev)
 	pci_save_state(pci);
 	pci_disable_device(pci);
 
-	dev_dbg(dev, "suspend done\n");
 	return 0;
 }
 
@@ -2232,7 +2205,6 @@ static int snd_echo_resume(struct device *dev)
 	u32 pipe_alloc_mask;
 	int err;
 
-	dev_dbg(dev, "resume start\n");
 	pci_restore_state(pci);
 	commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL);
 	if (commpage_bak == NULL)
@@ -2247,7 +2219,6 @@ static int snd_echo_resume(struct device *dev)
 		snd_echo_free(chip);
 		return err;
 	}
-	dev_dbg(dev, "resume init OK\n");
 
 	/* Temporarily set chip->pipe_alloc_mask=0 otherwise
 	 * restore_dsp_settings() fails.
@@ -2260,7 +2231,6 @@ static int snd_echo_resume(struct device *dev)
 		kfree(commpage_bak);
 		return err;
 	}
-	dev_dbg(dev, "resume restore OK\n");
 
 	memcpy(&commpage->audio_format, &commpage_bak->audio_format,
 		sizeof(commpage->audio_format));
@@ -2286,7 +2256,6 @@ static int snd_echo_resume(struct device *dev)
 		snd_echo_midi_output_trigger(chip->midi_out, 1);
 #endif
 
-	dev_dbg(dev, "resume done\n");
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
index 50a21690447d..2fa66dc3e675 100644
--- a/sound/pci/echoaudio/echoaudio_3g.c
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
 	u32 control_reg, clocks_from_dsp;
 
-	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	/* Mask off the clock select bits */
 	control_reg = le32_to_cpu(chip->comm_page->control_register) &
@@ -337,13 +336,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
 		if (chip->digital_mode == DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n");
 		control_reg |= E3G_SPDIF_CLOCK;
 		if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96)
 			control_reg |= E3G_DOUBLE_SPEED_MODE;
@@ -353,12 +350,10 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	case ECHO_CLOCK_ADAT:
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n");
 		control_reg |= E3G_ADAT_CLOCK;
 		control_reg &= ~E3G_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_WORD:
-		dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n");
 		control_reg |= E3G_WORD_CLOCK;
 		if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96)
 			control_reg |= E3G_DOUBLE_SPEED_MODE;
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index ba9d4f16cbb3..1a9427aabe1d 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic)
 			goto la_error;
 	}
 
-	dev_dbg(chip->card->dev, "ASIC loaded\n");
 	free_firmware(fw, chip);
 	return 0;
 
@@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code)
 
 			chip->dsp_code = code;		/* Show which DSP code loaded */
 			chip->bad_board = FALSE;	/* DSP OK */
-			dev_dbg(chip->card->dev, "load_dsp: OK!\n");
 			return 0;
 		}
 		udelay(100);
@@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters)
 static int restore_dsp_rettings(struct echoaudio *chip)
 {
 	int i, o, err;
-	dev_dbg(chip->card->dev, "restore_dsp_settings\n");
 
 	if ((err = check_asic_status(chip)) < 0)
 		return err;
@@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip)
 	if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0)
 		return -EIO;
 
-	dev_dbg(chip->card->dev, "restore_dsp_rettings done\n");
 	return 0;
 }
 
@@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */
 static int start_transport(struct echoaudio *chip, u32 channel_mask,
 			   u32 cyclic_mask)
 {
-	dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask,
 
 static int pause_transport(struct echoaudio *chip, u32 channel_mask)
 {
-	dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask)
 
 static int stop_transport(struct echoaudio *chip, u32 channel_mask)
 {
-	dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask);
 
 	if (wait_handshake(chip))
 		return -EIO;
@@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index)
 stopped and unallocated. */
 static int rest_in_peace(struct echoaudio *chip)
 {
-	dev_dbg(chip->card->dev,
-		"rest_in_peace() open=%x\n", chip->pipe_alloc_mask);
 
 	/* Stops all active pipes (just to be sure) */
 	stop_transport(chip, chip->active_mask);
@@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip)
  */
 static int init_line_levels(struct echoaudio *chip)
 {
-	dev_dbg(chip->card->dev, "init_line_levels\n");
 	memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain));
 	memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain));
 	memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain));
@@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
 	it moves data. The DMA counter is in units of bytes, not samples. */
 	pipe->dma_counter = &chip->comm_page->position[pipe_index];
 	*pipe->dma_counter = 0;
-	dev_dbg(chip->card->dev, "allocate_pipes: ok\n");
 	return pipe_index;
 }
 
@@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe)
 	u32 channel_mask;
 	int i;
 
-	dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index);
 	if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index)))
 		return -EINVAL;
 	if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED))
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
index a959eae95e0d..5dafe9260cb4 100644
--- a/sound/pci/echoaudio/gina20_dsp.c
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Gina20\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20))
 		return -ENODEV;
 
@@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate)
 
 static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
-	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
@@ -159,7 +156,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
 		set_sample_rate(chip, chip->sample_rate);
 		chip->input_clock = clock;
-		dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n");
 		break;
 	case ECHO_CLOCK_SPDIF:
 		chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN;
@@ -167,7 +163,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		clear_handshake(chip);
 		send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
 		chip->clock_state = GD_CLOCK_SPDIFIN;
-		dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n");
 		chip->input_clock = clock;
 		break;
 	default:
@@ -209,7 +204,6 @@ static int update_flags(struct echoaudio *chip)
 
 static int set_professional_spdif(struct echoaudio *chip, char prof)
 {
-	dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof);
 	if (prof)
 		chip->comm_page->flags |=
 			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c
index c8ea57612d22..6971766938bf 100644
--- a/sound/pci/echoaudio/gina24_dsp.c
+++ b/sound/pci/echoaudio/gina24_dsp.c
@@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Gina24\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24))
 		return -ENODEV;
 
@@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip)
 		control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
 		err = write_control_reg(chip, control_reg, TRUE);
 	}
-	dev_dbg(chip->card->dev, "load_asic() done\n");
 	return err;
 }
 
@@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 {
 	u32 control_reg, clocks_from_dsp;
 
-	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	/* Mask off the clock select bits */
 	control_reg = le32_to_cpu(chip->comm_page->control_register) &
@@ -247,13 +243,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
 		if (chip->digital_mode == DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n");
 		control_reg |= GML_SPDIF_CLOCK;
 		if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
 			control_reg |= GML_DOUBLE_SPEED_MODE;
@@ -263,17 +257,14 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	case ECHO_CLOCK_ADAT:
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
-		dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n");
 		control_reg |= GML_ADAT_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_ESYNC:
-		dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n");
 		control_reg |= GML_ESYNC_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_ESYNC96:
-		dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n");
 		control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE;
 		break;
 	default:
diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
index cdeb073fad85..54edd67edff7 100644
--- a/sound/pci/echoaudio/indigo_dsp.c
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Indigo\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO))
 		return -ENODEV;
 
@@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
index 133915ca6438..2cf5cc092d7a 100644
--- a/sound/pci/echoaudio/indigodj_dsp.c
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ))
 		return -ENODEV;
 
@@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c
index 26cdfcfc5553..5252863f1681 100644
--- a/sound/pci/echoaudio/indigodjx_dsp.c
+++ b/sound/pci/echoaudio/indigodjx_dsp.c
@@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX))
 		return -ENODEV;
 
@@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
index 5e6df7c25055..4e81787627e0 100644
--- a/sound/pci/echoaudio/indigoio_dsp.c
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO))
 		return -ENODEV;
 
@@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c
index 90cdd271d9fc..6de3f9bc6d26 100644
--- a/sound/pci/echoaudio/indigoiox_dsp.c
+++ b/sound/pci/echoaudio/indigoiox_dsp.c
@@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX))
 		return -ENODEV;
 
@@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
index 7f0f6ea08ca7..f2024a3565af 100644
--- a/sound/pci/echoaudio/layla20_dsp.c
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Layla20\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20))
 		return -ENODEV;
 
@@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -190,24 +188,19 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source)
 	u16 clock;
 	u32 rate;
 
-	dev_dbg(chip->card->dev, "set_input_clock:\n");
 	rate = 0;
 	switch (clock_source) {
 	case ECHO_CLOCK_INTERNAL:
-		dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n");
 		rate = chip->sample_rate;
 		clock = LAYLA20_CLOCK_INTERNAL;
 		break;
 	case ECHO_CLOCK_SPDIF:
-		dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n");
 		clock = LAYLA20_CLOCK_SPDIF;
 		break;
 	case ECHO_CLOCK_WORD:
-		dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n");
 		clock = LAYLA20_CLOCK_WORD;
 		break;
 	case ECHO_CLOCK_SUPER:
-		dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n");
 		clock = LAYLA20_CLOCK_SUPER;
 		break;
 	default:
@@ -232,7 +225,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source)
 
 static int set_output_clock(struct echoaudio *chip, u16 clock)
 {
-	dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock);
 	switch (clock) {
 	case ECHO_CLOCK_SUPER:
 		clock = LAYLA20_OUTPUT_CLOCK_SUPER;
@@ -286,7 +278,6 @@ static int update_flags(struct echoaudio *chip)
 
 static int set_professional_spdif(struct echoaudio *chip, char prof)
 {
-	dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof);
 	if (prof)
 		chip->comm_page->flags |=
 			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c
index eb8f218f79fe..4f11e81f6c0e 100644
--- a/sound/pci/echoaudio/layla24_dsp.c
+++ b/sound/pci/echoaudio/layla24_dsp.c
@@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Layla24\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24))
 		return -ENODEV;
 
@@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip)
 	if (chip->asic_loaded)
 		return 1;
 
-	dev_dbg(chip->card->dev, "load_asic\n");
 
 	/* Give the DSP a few milliseconds to settle down */
 	mdelay(10);
@@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip)
 		err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ,
 					TRUE);
 	
-	dev_dbg(chip->card->dev, "load_asic() done\n");
 	return err;
 }
 
@@ -262,7 +258,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	/* Pick the new clock */
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
@@ -271,7 +266,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		control_reg |= GML_SPDIF_CLOCK;
 		/* Layla24 doesn't support 96KHz S/PDIF */
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
-		dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n");
 		break;
 	case ECHO_CLOCK_WORD:
 		control_reg |= GML_WORD_CLOCK;
@@ -279,14 +273,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 			control_reg |= GML_DOUBLE_SPEED_MODE;
 		else
 			control_reg &= ~GML_DOUBLE_SPEED_MODE;
-		dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n");
 		break;
 	case ECHO_CLOCK_ADAT:
 		if (chip->digital_mode != DIGITAL_MODE_ADAT)
 			return -EAGAIN;
 		control_reg |= GML_ADAT_CLOCK;
 		control_reg &= ~GML_DOUBLE_SPEED_MODE;
-		dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n");
 		break;
 	default:
 		dev_err(chip->card->dev,
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
index ed2f21dcd1c9..fdad079ad9a1 100644
--- a/sound/pci/echoaudio/mia_dsp.c
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Mia\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA))
 		return -ENODEV;
 
@@ -67,7 +66,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
index 8d43c5a4976b..d913749d154a 100644
--- a/sound/pci/echoaudio/midi.c
+++ b/sound/pci/echoaudio/midi.c
@@ -157,7 +157,6 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream)
 	struct echoaudio *chip = substream->rmidi->private_data;
 
 	chip->midi_in = substream;
-	dev_dbg(chip->card->dev, "rawmidi_iopen\n");
 	return 0;
 }
 
@@ -183,7 +182,6 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream)
 	struct echoaudio *chip = substream->rmidi->private_data;
 
 	chip->midi_in = NULL;
-	dev_dbg(chip->card->dev, "rawmidi_iclose\n");
 	return 0;
 }
 
@@ -196,7 +194,6 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream)
 	chip->tinuse = 0;
 	chip->midi_full = 0;
 	chip->midi_out = substream;
-	dev_dbg(chip->card->dev, "rawmidi_open\n");
 	return 0;
 }
 
@@ -209,7 +206,6 @@ static void snd_echo_midi_output_write(unsigned long data)
 	int bytes, sent, time;
 	unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1];
 
-	dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n");
 	/* No interrupts are involved: we have to check at regular intervals
 	if the card's output buffer has room for new data. */
 	sent = bytes = 0;
@@ -288,7 +284,6 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream)
 	struct echoaudio *chip = substream->rmidi->private_data;
 
 	chip->midi_out = NULL;
-	dev_dbg(chip->card->dev, "rawmidi_oclose\n");
 	return 0;
 }
 
@@ -328,6 +323,5 @@ static int snd_echo_midi_create(struct snd_card *card,
 
 	chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
 		SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
-	dev_dbg(chip->card->dev, "MIDI ok\n");
 	return 0;
 }
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
index cc46a8c8e3dd..843c7a5e5105 100644
--- a/sound/pci/echoaudio/mona_dsp.c
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 {
 	int err;
 
-	dev_dbg(chip->card->dev, "init_hw() - Mona\n");
 	if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA))
 		return -ENODEV;
 
@@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 		return err;
 	chip->bad_board = FALSE;
 
-	dev_dbg(chip->card->dev, "init_hw done\n");
 	return err;
 }
 
@@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	u32 control_reg, clocks_from_dsp;
 	int err;
 
-	dev_dbg(chip->card->dev, "set_input_clock:\n");
 
 	/* Prevent two simultaneous calls to switch_asic() */
 	if (atomic_read(&chip->opencount))
@@ -315,7 +312,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 
 	switch (clock) {
 	case ECHO_CLOCK_INTERNAL:
-		dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n");
 		chip->input_clock = ECHO_CLOCK_INTERNAL;
 		return set_sample_rate(chip, chip->sample_rate);
 	case ECHO_CLOCK_SPDIF:
@@ -327,7 +323,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 		spin_lock_irq(&chip->lock);
 		if (err < 0)
 			return err;
-		dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n");
 		control_reg |= GML_SPDIF_CLOCK;
 		if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
 			control_reg |= GML_DOUBLE_SPEED_MODE;
@@ -335,7 +330,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 			control_reg &= ~GML_DOUBLE_SPEED_MODE;
 		break;
 	case ECHO_CLOCK_WORD:
-		dev_dbg(chip->card->dev, "Set Mona clock to WORD\n");
 		spin_unlock_irq(&chip->lock);
 		err = switch_asic(chip, clocks_from_dsp &
 				  GML_CLOCK_DETECT_BIT_WORD96);
-- 
cgit v1.2.3


From 1fb8510cdb5b7befe8a59f533c7fc12ef0dac73e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 7 Nov 2014 17:08:28 +0100
Subject: ALSA: pcm: Add snd_pcm_stop_xrun() helper

Add a new helper function snd_pcm_stop_xrun() to the standard sequnce
lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the
existing open codes with this helper.

The function checks the PCM running state to prevent setting the wrong
state, too, for more safety.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 drivers/media/pci/saa7134/saa7134-alsa.c |  4 +---
 include/sound/pcm.h                      |  1 +
 sound/arm/pxa2xx-pcm-lib.c               |  4 +---
 sound/core/pcm_native.c                  | 23 +++++++++++++++++++++++
 sound/firewire/amdtp.c                   |  8 ++------
 sound/firewire/isight.c                  | 10 ++--------
 sound/pci/asihpi/asihpi.c                |  5 +----
 sound/pci/atiixp.c                       |  4 +---
 sound/pci/atiixp_modem.c                 |  4 +---
 sound/soc/atmel/atmel-pcm-dma.c          |  4 +---
 sound/soc/fsl/fsl_dma.c                  |  9 +--------
 sound/usb/6fire/pcm.c                    | 17 ++++-------------
 sound/usb/endpoint.c                     |  4 +---
 sound/usb/misc/ua101.c                   | 18 ++++--------------
 sound/usb/usx2y/usbusx2yaudio.c          |  9 ++-------
 15 files changed, 46 insertions(+), 78 deletions(-)

(limited to 'sound/pci')

diff --git a/drivers/media/pci/saa7134/saa7134-alsa.c b/drivers/media/pci/saa7134/saa7134-alsa.c
index 40569894c1c9..ac3cd74e824e 100644
--- a/drivers/media/pci/saa7134/saa7134-alsa.c
+++ b/drivers/media/pci/saa7134/saa7134-alsa.c
@@ -173,9 +173,7 @@ static void saa7134_irq_alsa_done(struct saa7134_dev *dev,
 		dprintk("irq: overrun [full=%d/%d] - Blocks in %d\n",dev->dmasound.read_count,
 			dev->dmasound.bufsize, dev->dmasound.blocks);
 		spin_unlock(&dev->slock);
-		snd_pcm_stream_lock(dev->dmasound.substream);
-		snd_pcm_stop(dev->dmasound.substream,SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock(dev->dmasound.substream);
+		snd_pcm_stop_xrun(dev->dmasound.substream);
 		return;
 	}
 
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0b8daeed0a33..40289ec2451c 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -506,6 +506,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
 int snd_pcm_start(struct snd_pcm_substream *substream);
 int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status);
 int snd_pcm_drain_done(struct snd_pcm_substream *substream);
+int snd_pcm_stop_xrun(struct snd_pcm_substream *substream);
 #ifdef CONFIG_PM
 int snd_pcm_suspend(struct snd_pcm_substream *substream);
 int snd_pcm_suspend_all(struct snd_pcm *pcm);
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index a61d7a9a995e..01f8fdc42b1b 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -200,9 +200,7 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
 	} else {
 		printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
 			dma_ch, dcsr);
-		snd_pcm_stream_lock(substream);
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock(substream);
+		snd_pcm_stop_xrun(substream);
 	}
 }
 EXPORT_SYMBOL(pxa2xx_pcm_dma_irq);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index dfb5031969f8..a3d122109704 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1098,6 +1098,29 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream)
 				     SNDRV_PCM_STATE_SETUP);
 }
 
+/**
+ * snd_pcm_stop_xrun - stop the running streams as XRUN
+ * @substream: the PCM substream instance
+ * @state: PCM state after stopping the stream
+ *
+ * This stops the given running substream (and all linked substreams) as XRUN.
+ * Unlike snd_pcm_stop(), this function takes the substream lock by itself.
+ *
+ * Return: Zero if successful, or a negative error code.
+ */
+int snd_pcm_stop_xrun(struct snd_pcm_substream *substream)
+{
+	unsigned long flags;
+	int ret = 0;
+
+	snd_pcm_stream_lock_irqsave(substream, flags);
+	if (snd_pcm_running(substream))
+		ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+	snd_pcm_stream_unlock_irqrestore(substream, flags);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun);
+
 /*
  * pause callbacks
  */
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 95fc2eaf11dc..3badc70124ab 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -1006,11 +1006,7 @@ void amdtp_stream_pcm_abort(struct amdtp_stream *s)
 	struct snd_pcm_substream *pcm;
 
 	pcm = ACCESS_ONCE(s->pcm);
-	if (pcm) {
-		snd_pcm_stream_lock_irq(pcm);
-		if (snd_pcm_running(pcm))
-			snd_pcm_stop(pcm, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock_irq(pcm);
-	}
+	if (pcm)
+		snd_pcm_stop_xrun(pcm);
 }
 EXPORT_SYMBOL(amdtp_stream_pcm_abort);
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 7ac94439e758..48d6dca471c6 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -131,14 +131,8 @@ static void isight_samples(struct isight *isight,
 
 static void isight_pcm_abort(struct isight *isight)
 {
-	unsigned long flags;
-
-	if (ACCESS_ONCE(isight->pcm_active)) {
-		snd_pcm_stream_lock_irqsave(isight->pcm, flags);
-		if (snd_pcm_running(isight->pcm))
-			snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock_irqrestore(isight->pcm, flags);
-	}
+	if (ACCESS_ONCE(isight->pcm_active))
+		snd_pcm_stop_xrun(isight->pcm);
 }
 
 static void isight_dropped_samples(struct isight *isight, unsigned int total)
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index ac66b3228a34..ff9f9f1c0391 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -769,10 +769,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 						s->number);
 				ds->drained_count++;
 				if (ds->drained_count > 20) {
-					unsigned long flags;
-					snd_pcm_stream_lock_irqsave(s, flags);
-					snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN);
-					snd_pcm_stream_unlock_irqrestore(s, flags);
+					snd_pcm_stop_xrun(s);
 					continue;
 				}
 			} else {
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 7895c5d300c7..9c1c4452a8ee 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -688,9 +688,7 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma)
 	if (! dma->substream || ! dma->running)
 		return;
 	dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type);
-	snd_pcm_stream_lock(dma->substream);
-	snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN);
-	snd_pcm_stream_unlock(dma->substream);
+	snd_pcm_stop_xrun(dma->substream);
 }
 
 /*
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 3c3241309a30..b2f63e0727de 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -638,9 +638,7 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip,
 	if (! dma->substream || ! dma->running)
 		return;
 	dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type);
-	snd_pcm_stream_lock(dma->substream);
-	snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN);
-	snd_pcm_stream_unlock(dma->substream);
+	snd_pcm_stop_xrun(dma->substream);
 }
 
 /*
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index b79a2a864513..33fb3bb133df 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -80,9 +80,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
 
 		/* stop RX and capture: will be enabled again at restart */
 		ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable);
-		snd_pcm_stream_lock(substream);
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock(substream);
+		snd_pcm_stop_xrun(substream);
 
 		/* now drain RHR and read status to remove xrun condition */
 		ssc_readx(prtd->ssc->regs, SSC_RHR);
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index a609aafc994d..b2b108805b24 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -151,14 +151,7 @@ static const struct snd_pcm_hardware fsl_dma_hardware = {
  */
 static void fsl_dma_abort_stream(struct snd_pcm_substream *substream)
 {
-	unsigned long flags;
-
-	snd_pcm_stream_lock_irqsave(substream, flags);
-
-	if (snd_pcm_running(substream))
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-
-	snd_pcm_stream_unlock_irqrestore(substream, flags);
+	snd_pcm_stop_xrun(substream);
 }
 
 /**
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index ba40489b2de4..36f4115eb1cd 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -679,25 +679,16 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
 void usb6fire_pcm_abort(struct sfire_chip *chip)
 {
 	struct pcm_runtime *rt = chip->pcm;
-	unsigned long flags;
 	int i;
 
 	if (rt) {
 		rt->panic = true;
 
-		if (rt->playback.instance) {
-			snd_pcm_stream_lock_irqsave(rt->playback.instance, flags);
-			snd_pcm_stop(rt->playback.instance,
-					SNDRV_PCM_STATE_XRUN);
-			snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags);
-		}
+		if (rt->playback.instance)
+			snd_pcm_stop_xrun(rt->playback.instance);
 
-		if (rt->capture.instance) {
-			snd_pcm_stream_lock_irqsave(rt->capture.instance, flags);
-			snd_pcm_stop(rt->capture.instance,
-					SNDRV_PCM_STATE_XRUN);
-			snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags);
-		}
+		if (rt->capture.instance)
+			snd_pcm_stop_xrun(rt->capture.instance);
 
 		for (i = 0; i < PCM_N_URBS; i++) {
 			usb_poison_urb(&rt->in_urbs[i].instance);
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index a4679913b0aa..03b074419964 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -391,9 +391,7 @@ static void snd_complete_urb(struct urb *urb)
 	usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err);
 	if (ep->data_subs && ep->data_subs->pcm_substream) {
 		substream = ep->data_subs->pcm_substream;
-		snd_pcm_stream_lock_irqsave(substream, flags);
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock_irqrestore(substream, flags);
+		snd_pcm_stop_xrun(substream);
 	}
 
 exit_clear:
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index a1bab149df4d..9581089c28c5 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -613,24 +613,14 @@ static int start_usb_playback(struct ua101 *ua)
 
 static void abort_alsa_capture(struct ua101 *ua)
 {
-	unsigned long flags;
-
-	if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) {
-		snd_pcm_stream_lock_irqsave(ua->capture.substream, flags);
-		snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags);
-	}
+	if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states))
+		snd_pcm_stop_xrun(ua->capture.substream);
 }
 
 static void abort_alsa_playback(struct ua101 *ua)
 {
-	unsigned long flags;
-
-	if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) {
-		snd_pcm_stream_lock_irqsave(ua->playback.substream, flags);
-		snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags);
-	}
+	if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states))
+		snd_pcm_stop_xrun(ua->playback.substream);
 }
 
 static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream,
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index a63330dd1407..61d5dc2a3421 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -272,13 +272,8 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y)
 	for (s = 0; s < 4; s++) {
 		struct snd_usX2Y_substream *subs = usX2Y->subs[s];
 		if (subs) {
-			if (atomic_read(&subs->state) >= state_PRERUNNING) {
-				unsigned long flags;
-
-				snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags);
-				snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
-				snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags);
-			}
+			if (atomic_read(&subs->state) >= state_PRERUNNING)
+				snd_pcm_stop_xrun(subs->pcm_substream);
 			for (u = 0; u < NRURBS; u++) {
 				struct urb *urb = subs->urb[u];
 				if (NULL != urb)
-- 
cgit v1.2.3


From 2a9e8df00951092e825144a9968285398f8aa162 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Nov 2014 16:46:35 +0100
Subject: ALSA: vx: Fix missing kerneldoc parameter descriptions

The file isn't processed, but it's not bad to fix beforehand.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/vx/vx_core.c  | 10 ++++++++++
 sound/pci/vx222/vx222_ops.c |  5 +++++
 sound/pcmcia/vx/vxpocket.c  |  1 +
 3 files changed, 16 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index e8cc16993903..fc05a37fd017 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -416,6 +416,7 @@ int vx_send_rih(struct vx_core *chip, int cmd)
 
 /**
  * snd_vx_boot_xilinx - boot up the xilinx interface
+ * @chip: VX core instance
  * @boot: the boot record to load
  */
 int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *boot)
@@ -538,6 +539,8 @@ EXPORT_SYMBOL(snd_vx_threaded_irq_handler);
 
 /**
  * snd_vx_irq_handler - interrupt handler
+ * @irq: irq number
+ * @dev: VX core instance
  */
 irqreturn_t snd_vx_irq_handler(int irq, void *dev)
 {
@@ -649,6 +652,8 @@ static void vx_proc_init(struct vx_core *chip)
 
 /**
  * snd_vx_dsp_boot - load the DSP boot
+ * @chip: VX core instance
+ * @boot: firmware data
  */
 int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *boot)
 {
@@ -669,6 +674,8 @@ EXPORT_SYMBOL(snd_vx_dsp_boot);
 
 /**
  * snd_vx_dsp_load - load the DSP image
+ * @chip: VX core instance
+ * @dsp: firmware data
  */
 int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp)
 {
@@ -768,7 +775,10 @@ EXPORT_SYMBOL(snd_vx_resume);
 
 /**
  * snd_vx_create - constructor for struct vx_core
+ * @card: card instance
  * @hw: hardware specific record
+ * @ops: VX ops pointer
+ * @extra_size: extra byte size to allocate appending to chip
  *
  * this function allocates the instance and prepare for the hardware
  * initialization.
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index 2d1570273e99..52c1a8d5b88a 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -92,6 +92,7 @@ static inline unsigned long vx2_reg_addr(struct vx_core *_chip, int reg)
 
 /**
  * snd_vx_inb - read a byte from the register
+ * @chip: VX core instance
  * @offset: register enum
  */
 static unsigned char vx2_inb(struct vx_core *chip, int offset)
@@ -101,6 +102,7 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset)
 
 /**
  * snd_vx_outb - write a byte on the register
+ * @chip: VX core instance
  * @offset: the register offset
  * @val: the value to write
  */
@@ -114,6 +116,7 @@ static void vx2_outb(struct vx_core *chip, int offset, unsigned char val)
 
 /**
  * snd_vx_inl - read a 32bit word from the register
+ * @chip: VX core instance
  * @offset: register enum
  */
 static unsigned int vx2_inl(struct vx_core *chip, int offset)
@@ -123,6 +126,7 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset)
 
 /**
  * snd_vx_outl - write a 32bit word on the register
+ * @chip: VX core instance
  * @offset: the register enum
  * @val: the value to write
  */
@@ -223,6 +227,7 @@ static int vx2_test_xilinx(struct vx_core *_chip)
 
 /**
  * vx_setup_pseudo_dma - set up the pseudo dma read/write mode.
+ * @chip: VX core instance
  * @do_write: 0 = read, 1 = set up for DMA write
  */
 static void vx2_setup_pseudo_dma(struct vx_core *chip, int do_write)
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 92ec11456e3a..b16f42deed67 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -174,6 +174,7 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl,
 
 /**
  * snd_vxpocket_assign_resources - initialize the hardware and card instance.
+ * @chip: VX core instance
  * @port: i/o port for the card
  * @irq: irq number for the card
  *
-- 
cgit v1.2.3


From e60b2c7fcdef03256cde864d678df240877a5e80 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Nov 2014 16:47:26 +0100
Subject: ALSA: hda - Fix kerneldoc errors in patch_ca0132.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 4f7ffa8c4a0d..e0383eea9880 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2417,7 +2417,7 @@ static int dspxfr_one_seg(struct hda_codec *codec,
  * @reloc: Relocation address for loading single-segment overlays, or 0 for
  *	   no relocation
  * @sample_rate: sampling rate of the stream used for DSP download
- * @number_channels: channels of the stream used for DSP download
+ * @channels: channels of the stream used for DSP download
  * @ovly: TRUE if overlay format is required
  *
  * Returns zero or a negative error code.
@@ -2556,10 +2556,7 @@ static void dspload_post_setup(struct hda_codec *codec)
 }
 
 /**
- * Download DSP from a DSP Image Fast Load structure. This structure is a
- * linear, non-constant sized element array of structures, each of which
- * contain the count of the data to be loaded, the data itself, and the
- * corresponding starting chip address of the starting data location.
+ * dspload_image - Download DSP from a DSP Image Fast Load structure.
  *
  * @codec: the HDA codec
  * @fls: pointer to a fast load image
@@ -2570,6 +2567,10 @@ static void dspload_post_setup(struct hda_codec *codec)
  * @router_chans: number of audio router channels to be allocated (0 means use
  *		  internal defaults; max is 32)
  *
+ * Download DSP from a DSP Image Fast Load structure. This structure is a
+ * linear, non-constant sized element array of structures, each of which
+ * contain the count of the data to be loaded, the data itself, and the
+ * corresponding starting chip address of the starting data location.
  * Returns zero or a negative error code.
  */
 static int dspload_image(struct hda_codec *codec,
-- 
cgit v1.2.3


From 3f60c87d129acbb232afbc7269c726d009a01869 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Nov 2014 17:09:22 +0100
Subject: ALSA: mixart: Fix kerneldoc comments

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/mixart/mixart_hwdep.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c
index 581e1e74863c..9996a4dead0f 100644
--- a/sound/pci/mixart/mixart_hwdep.c
+++ b/sound/pci/mixart/mixart_hwdep.c
@@ -37,10 +37,11 @@
 /**
  * wait for a value on a peudo register, exit with a timeout
  *
- * @param mgr pointer to miXart manager structure
- * @param offset unsigned pseudo_register base + offset of value
- * @param value value
- * @param timeout timeout in centisenconds
+ * @mgr: pointer to miXart manager structure
+ * @offset: unsigned pseudo_register base + offset of value
+ * @is_egal: wait for the equal value
+ * @value: value
+ * @timeout: timeout in centisenconds
  */
 static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr,
 					       u32 offset, int is_egal,
-- 
cgit v1.2.3


From ddcecf6b6ae7b91c8735e52f50cd403ee9cbe298 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Nov 2014 17:24:26 +0100
Subject: ALSA: Fix invalid kerneldoc markers

They are no real kerneldoc comments, so drop such markers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/uapi/sound/hdspm.h   | 12 +++++------
 sound/pci/emu10k1/emu10k1x.c |  2 +-
 sound/pci/lx6464es/lx_defs.h |  2 +-
 sound/pci/rme9652/hdspm.c    | 48 ++++++++++++++++++++++----------------------
 4 files changed, 32 insertions(+), 32 deletions(-)

(limited to 'sound/pci')

diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index d956c3593f65..b357f1a5e29c 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -74,14 +74,14 @@ struct hdspm_config {
 #define SNDRV_HDSPM_IOCTL_GET_CONFIG \
 	_IOR('H', 0x41, struct hdspm_config)
 
-/**
+/*
  * If there's a TCO (TimeCode Option) board installed,
  * there are further options and status data available.
  * The hdspm_ltc structure contains the current SMPTE
  * timecode and some status information and can be
  * obtained via SNDRV_HDSPM_IOCTL_GET_LTC or in the
  * hdspm_status struct.
- **/
+ */
 
 enum hdspm_ltc_format {
 	format_invalid,
@@ -113,11 +113,11 @@ struct hdspm_ltc {
 
 #define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc)
 
-/**
+/*
  * The status data reflects the device's current state
  * as determined by the card's configuration and
  * connection status.
- **/
+ */
 
 enum hdspm_sync {
 	hdspm_sync_no_lock = 0,
@@ -171,9 +171,9 @@ struct hdspm_status {
 #define SNDRV_HDSPM_IOCTL_GET_STATUS \
 	_IOR('H', 0x47, struct hdspm_status)
 
-/**
+/*
  * Get information about the card and its add-ons.
- **/
+ */
 
 #define HDSPM_ADDON_TCO 1
 
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index e223de1408c0..15933f92f63a 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -180,7 +180,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
 
 /* From 0x50 - 0x5f, last samples captured */
 
-/**
+/*
  * The hardware has 3 channels for playback and 1 for capture.
  *  - channel 0 is the front channel
  *  - channel 1 is the rear channel
diff --git a/sound/pci/lx6464es/lx_defs.h b/sound/pci/lx6464es/lx_defs.h
index 49d36bdd512c..469bcc685edf 100644
--- a/sound/pci/lx6464es/lx_defs.h
+++ b/sound/pci/lx6464es/lx_defs.h
@@ -175,7 +175,7 @@ enum buffer_flags {
 	BF_ZERO		= 0x00,	/* no flags (init).*/
 };
 
-/**
+/*
 *	Stream Flags definitions
 */
 enum stream_flags {
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index e09348c156d8..3342705a5715 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -2201,10 +2201,10 @@ static inline int hdspm_get_pll_freq(struct hdspm *hdspm)
 	return rate;
 }
 
-/**
+/*
  * Calculate the real sample rate from the
  * current DDS value.
- **/
+ */
 static int hdspm_get_system_sample_rate(struct hdspm *hdspm)
 {
 	unsigned int rate;
@@ -2270,9 +2270,9 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol,
 }
 
 
-/**
+/*
  * Returns the WordClock sample rate class for the given card.
- **/
+ */
 static int hdspm_get_wc_sample_rate(struct hdspm *hdspm)
 {
 	int status;
@@ -2295,9 +2295,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm)
 }
 
 
-/**
+/*
  * Returns the TCO sample rate class for the given card.
- **/
+ */
 static int hdspm_get_tco_sample_rate(struct hdspm *hdspm)
 {
 	int status;
@@ -2321,9 +2321,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm)
 }
 
 
-/**
+/*
  * Returns the SYNC_IN sample rate class for the given card.
- **/
+ */
 static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm)
 {
 	int status;
@@ -2343,9 +2343,9 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm)
 	return 0;
 }
 
-/**
+/*
  * Returns the AES sample rate class for the given card.
- **/
+ */
 static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
 {
 	int timecode;
@@ -2361,10 +2361,10 @@ static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
 	return 0;
 }
 
-/**
+/*
  * Returns the sample rate class for input source <idx> for
  * 'new style' cards like the AIO and RayDAT.
- **/
+ */
 static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
 {
 	int status = hdspm_read(hdspm, HDSPM_RD_STATUS_2);
@@ -2512,10 +2512,10 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 }
 
 
-/**
+/*
  * Returns the system clock mode for the given card.
  * @returns 0 - master, 1 - slave
- **/
+ */
 static int hdspm_system_clock_mode(struct hdspm *hdspm)
 {
 	switch (hdspm->io_type) {
@@ -2534,10 +2534,10 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm)
 }
 
 
-/**
+/*
  * Sets the system clock mode.
  * @param mode 0 - master, 1 - slave
- **/
+ */
 static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode)
 {
 	hdspm_set_toggle_setting(hdspm,
@@ -2692,11 +2692,11 @@ static int snd_hdspm_put_clock_source(struct snd_kcontrol *kcontrol,
 }
 
 
-/**
+/*
  * Returns the current preferred sync reference setting.
  * The semantics of the return value are depending on the
  * card, please see the comments for clarification.
- **/
+ */
 static int hdspm_pref_sync_ref(struct hdspm * hdspm)
 {
 	switch (hdspm->io_type) {
@@ -2795,11 +2795,11 @@ static int hdspm_pref_sync_ref(struct hdspm * hdspm)
 }
 
 
-/**
+/*
  * Set the preferred sync reference to <pref>. The semantics
  * of <pref> are depending on the card type, see the comments
  * for clarification.
- **/
+ */
 static int hdspm_set_pref_sync_ref(struct hdspm * hdspm, int pref)
 {
 	int p = 0;
@@ -4101,9 +4101,9 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
 
 
 
-/**
+/*
  * TCO controls
- **/
+ */
 static void hdspm_tco_write(struct hdspm *hdspm)
 {
 	unsigned int tc[4] = { 0, 0, 0, 0};
@@ -5403,7 +5403,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
 			HDSPM_midi2IRQPending | HDSPM_midi3IRQPending);
 
 	/* now = get_cycles(); */
-	/**
+	/*
 	 *   LAT_2..LAT_0 period  counter (win)  counter (mac)
 	 *          6       4096   ~256053425     ~514672358
 	 *          5       2048   ~128024983     ~257373821
@@ -5412,7 +5412,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
 	 *          2        256    ~16003039      ~32260176
 	 *          1        128     ~7998738      ~16194507
 	 *          0         64     ~3998231       ~8191558
-	 **/
+	 */
 	/*
 	  dev_info(hdspm->card->dev, "snd_hdspm_interrupt %llu @ %llx\n",
 	   now-hdspm->last_interrupt, status & 0xFFC0);
-- 
cgit v1.2.3


From 387417b56295ef93d7cb38e1721826c85dfe897c Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudip@vectorindia.org>
Date: Fri, 14 Nov 2014 16:09:05 +0530
Subject: ALSA: ice1712: remove unneeded return statement

the functions:
        snd_ice1712_akm4xxx_build_controls
        snd_ice1712_build_pro_mixer
        snd_ctl_add
        snd_ak4114_build
        prodigy192_ak4114_init
        snd_ak4113_build
are all returning either 0 or a negetive error value.
so we can easily remove the check for a negative value and return
the value instead.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/hoontech.c   |  6 +-----
 sound/pci/ice1712/ice1712.c    | 19 ++++++-------------
 sound/pci/ice1712/ice1724.c    |  7 ++-----
 sound/pci/ice1712/juli.c       |  5 +----
 sound/pci/ice1712/prodigy192.c |  4 +---
 sound/pci/ice1712/quartet.c    |  5 +----
 6 files changed, 12 insertions(+), 34 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c
index 59e37c581691..a40001c1d9e8 100644
--- a/sound/pci/ice1712/hoontech.c
+++ b/sound/pci/ice1712/hoontech.c
@@ -309,11 +309,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice)
 		return err;
 
 	/* ak4524 controls */
-	err = snd_ice1712_akm4xxx_build_controls(ice);
-	if (err < 0)
-		return err;
-
-	return 0;
+	return snd_ice1712_akm4xxx_build_controls(ice);
 }
 
 static int snd_ice1712_ez8_init(struct snd_ice1712 *ice)
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 48a0c330da24..597533490f2d 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -1295,10 +1295,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd
 			return err;
 	}
 
-	err = snd_ice1712_build_pro_mixer(ice);
-	if (err < 0)
-		return err;
-	return 0;
+	return snd_ice1712_build_pro_mixer(ice);
 }
 
 /*
@@ -1545,10 +1542,9 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice)
 			dev_warn(ice->card->dev,
 				 "cannot initialize ac97 for consumer, skipped\n");
 		else {
-			err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice));
-			if (err < 0)
-				return err;
-			return 0;
+			return snd_ctl_add(ice->card,
+			snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97,
+				     ice));
 		}
 	}
 
@@ -2497,11 +2493,8 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice)
 	err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_volume_rate, ice));
 	if (err < 0)
 		return err;
-	err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice));
-	if (err < 0)
-		return err;
-
-	return 0;
+	return snd_ctl_add(ice->card,
+			   snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice));
 }
 
 static int snd_ice1712_free(struct snd_ice1712 *ice)
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index f633e3bb4c43..ea53167081b8 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2497,11 +2497,8 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice)
 			return err;
 	}
 
-	err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice));
-	if (err < 0)
-		return err;
-
-	return 0;
+	return snd_ctl_add(ice->card,
+			   snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice));
 }
 
 static int snd_vt1724_free(struct snd_ice1712 *ice)
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 7a6c0786c55c..a1536c1a7ed4 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -475,11 +475,8 @@ static int juli_add_controls(struct snd_ice1712 *ice)
 		return err;
 
 	/* only capture SPDIF over AK4114 */
-	err = snd_ak4114_build(spec->ak4114, NULL,
+	return snd_ak4114_build(spec->ak4114, NULL,
 			ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
-	if (err < 0)
-		return err;
-	return 0;
 }
 
 /*
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 1eb151aaa965..3919aed39ca0 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice)
 			"AK4114 initialized with status %d\n", err);
 	} else
 		dev_dbg(ice->card->dev, "AK4114 not found\n");
-	if (err < 0)
-		return err;
 
-	return 0;
+	return err;
 }
 
 
diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c
index d4caf9d05922..6f55e02e5c84 100644
--- a/sound/pci/ice1712/quartet.c
+++ b/sound/pci/ice1712/quartet.c
@@ -833,11 +833,8 @@ static int qtet_add_controls(struct snd_ice1712 *ice)
 	if (err < 0)
 		return err;
 	/* only capture SPDIF over AK4113 */
-	err = snd_ak4113_build(spec->ak4113,
+	return snd_ak4113_build(spec->ak4113,
 			ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
-	if (err < 0)
-		return err;
-	return 0;
 }
 
 static inline int qtet_is_spdif_master(struct snd_ice1712 *ice)
-- 
cgit v1.2.3


From b393df0145e271724fee10f93c023662f8557bb9 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudip@vectorindia.org>
Date: Fri, 14 Nov 2014 16:09:07 +0530
Subject: ALSA: ice1712: remove unused variable

buf_size was initialized with snd_pcm_lib_buffer_bytes,
but never used. and so it is safe to be deleted.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1712.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 597533490f2d..65251911cf6f 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -620,10 +620,9 @@ static int snd_ice1712_playback_ds_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	u32 period_size, buf_size, rate, tmp, chn;
+	u32 period_size, rate, tmp, chn;
 
 	period_size = snd_pcm_lib_period_bytes(substream) - 1;
-	buf_size = snd_pcm_lib_buffer_bytes(substream) - 1;
 	tmp = 0x0064;
 	if (snd_pcm_format_width(runtime->format) == 16)
 		tmp &= ~0x04;
-- 
cgit v1.2.3


From b8eca77e54525c818f35f51afb64fc13205443a3 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee <sudip@vectorindia.org>
Date: Fri, 14 Nov 2014 18:12:21 +0530
Subject: ALSA: ice1712: consider error value

earlier we were ignoring the return value of snd_ak4114_create and
always returning 0.
now we are returning the actual status. revo_init is calling this
function, and revo_init is checking the return value.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/revo.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index 1112ec1953be..1d81ae677573 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -494,11 +494,13 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice)
 				 ap192_ak4114_write,
 				 ak4114_init_vals, ak4114_init_txcsb,
 				 ice, &spec->ak4114);
+	if (err < 0)
+		return err;
 	/* AK4114 in Revo cannot detect external rate correctly.
 	 * No reason to stop capture stream due to incorrect checks */
 	spec->ak4114->check_flags = AK4114_CHECK_NO_RATE;
 
-	return 0; /* error ignored; it's no fatal error */
+	return 0;
 }
 
 static int revo_init(struct snd_ice1712 *ice)
-- 
cgit v1.2.3


From f0acd28c87ad2a5d1b40403fdd5defda2961b2a1 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Mon, 17 Nov 2014 10:44:33 +0100
Subject: ALSA: hda: Deletion of unnecessary checks before two function calls

The functions kfree() and release_firmware() test whether their argument
is NULL and then return immediately. Thus the test around the call
is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 3 +--
 sound/pci/hda/hda_intel.c | 3 +--
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index ca98f5209f8f..b2d58998dbdd 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -834,8 +834,7 @@ static void snd_hda_bus_free(struct hda_bus *bus)
 	WARN_ON(!list_empty(&bus->codec_list));
 	if (bus->workq)
 		flush_workqueue(bus->workq);
-	if (bus->unsol)
-		kfree(bus->unsol);
+	kfree(bus->unsol);
 	if (bus->ops.private_free)
 		bus->ops.private_free(bus);
 	if (bus->workq)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9ab1e631cb32..91fa959d05fe 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1130,8 +1130,7 @@ static int azx_free(struct azx *chip)
 	pci_disable_device(chip->pci);
 	kfree(chip->azx_dev);
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
-	if (chip->fw)
-		release_firmware(chip->fw);
+	release_firmware(chip->fw);
 #endif
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
 		hda_display_power(false);
-- 
cgit v1.2.3


From ae1b22658e6d3ebc6af07a225c221d84fe8cb91f Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Mon, 17 Nov 2014 11:28:02 +0100
Subject: ALSA: ice17xx: Deletion of unnecessary checks before the function
 call "snd_ac97_resume"

The snd_ac97_resume() function tests whether its argument is NULL and then
returns immediately. Thus the test around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1712.c | 3 +--
 sound/pci/ice1712/ice1724.c | 3 +--
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 65251911cf6f..b039b46152c6 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2878,8 +2878,7 @@ static int snd_ice1712_resume(struct device *dev)
 	outw(ice->pm_saved_spdif_ctrl, ICEMT(ice, ROUTE_SPDOUT));
 	outw(ice->pm_saved_route, ICEMT(ice, ROUTE_PSDOUT03));
 
-	if (ice->ac97)
-		snd_ac97_resume(ice->ac97);
+	snd_ac97_resume(ice->ac97);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index ea53167081b8..d73da157ea14 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2875,8 +2875,7 @@ static int snd_vt1724_resume(struct device *dev)
 	outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG));
 	outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK));
 
-	if (ice->ac97)
-		snd_ac97_resume(ice->ac97);
+	snd_ac97_resume(ice->ac97);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
-- 
cgit v1.2.3


From 6da95e1ea8e2492530eac9c51b293226e3f4ce94 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Mon, 17 Nov 2014 12:42:16 +0100
Subject: ALSA: lola: Deletion of an unnecessary check before the function call
 "vfree"

The vfree() function performs also input parameter validation. Thus the test
around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/lola/lola_mixer.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c
index 782f4d8299ae..e7fe15dd5a90 100644
--- a/sound/pci/lola/lola_mixer.c
+++ b/sound/pci/lola/lola_mixer.c
@@ -108,8 +108,7 @@ int lola_init_pins(struct lola *chip, int dir, int *nidp)
 
 void lola_free_mixer(struct lola *chip)
 {
-	if (chip->mixer.array_saved)
-		vfree(chip->mixer.array_saved);
+	vfree(chip->mixer.array_saved);
 }
 
 int lola_init_mixer_widget(struct lola *chip, int nid)
-- 
cgit v1.2.3


From c283661018e347bc72633969411974df8ec2ac92 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Mon, 17 Nov 2014 13:04:14 +0100
Subject: ALSA: hdsp: Deletion of an unnecessary check before the function call
 "release_firmware"

The release_firmware() function tests whether its argument is NULL and then
return immediately. Thus the test around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 2eb8baf7b828..cf5a6c8b9a63 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5307,8 +5307,7 @@ static int snd_hdsp_free(struct hdsp *hdsp)
 
 	snd_hdsp_free_buffers(hdsp);
 
-	if (hdsp->firmware)
-		release_firmware(hdsp->firmware);
+	release_firmware(hdsp->firmware);
 	vfree(hdsp->fw_uploaded);
 
 	if (hdsp->iobase)
-- 
cgit v1.2.3


From 0f32fd1900e6b972f289416dbd75e92772b630cb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Nov 2014 12:16:14 +0100
Subject: ALSA: hda/realtek - Clean up mute/mic GPIO LED handling

There are a few duplicated codes handling the mute and mic-mute LEDs
via GPIO pins.  Let's consolidate to single helpers.  Here we
introduced two new fields to alc_spec, gpio_mute_led_mask and
gpio_mic_led_mask, to contain the bit mask to set/clear.  Also,
mute_led_polarity is evaluated as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 81 +++++++++++++++++++++----------------------
 1 file changed, 40 insertions(+), 41 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1af917f58a70..3c29a558e7db 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -96,6 +96,8 @@ struct alc_spec {
 	hda_nid_t cap_mute_led_nid;
 
 	unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */
+	unsigned int gpio_mute_led_mask;
+	unsigned int gpio_mic_led_mask;
 
 	hda_nid_t headset_mic_pin;
 	hda_nid_t headphone_mic_pin;
@@ -3235,41 +3237,45 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec,
 	}
 }
 
-/* turn on/off mute LED per vmaster hook */
-static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled)
+/* update LED status via GPIO */
+static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
+				bool enabled)
 {
-	struct hda_codec *codec = private_data;
 	struct alc_spec *spec = codec->spec;
 	unsigned int oldval = spec->gpio_led;
 
+	if (spec->mute_led_polarity)
+		enabled = !enabled;
+
 	if (enabled)
-		spec->gpio_led &= ~0x08;
+		spec->gpio_led &= ~mask;
 	else
-		spec->gpio_led |= 0x08;
+		spec->gpio_led |= mask;
 	if (spec->gpio_led != oldval)
 		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
 				    spec->gpio_led);
 }
 
-/* turn on/off mic-mute LED per capture hook */
-static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec,
-					       struct snd_kcontrol *kcontrol,
-					       struct snd_ctl_elem_value *ucontrol)
+/* turn on/off mute LED via GPIO per vmaster hook */
+static void alc_fixup_gpio_mute_hook(void *private_data, int enabled)
 {
+	struct hda_codec *codec = private_data;
 	struct alc_spec *spec = codec->spec;
-	unsigned int oldval = spec->gpio_led;
 
-	if (!ucontrol)
-		return;
+	alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+}
 
-	if (ucontrol->value.integer.value[0] ||
-	    ucontrol->value.integer.value[1])
-		spec->gpio_led &= ~0x10;
-	else
-		spec->gpio_led |= 0x10;
-	if (spec->gpio_led != oldval)
-		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-				    spec->gpio_led);
+/* turn on/off mic-mute LED via GPIO per capture hook */
+static void alc_fixup_gpio_mic_mute_hook(struct hda_codec *codec,
+					 struct snd_kcontrol *kcontrol,
+					 struct snd_ctl_elem_value *ucontrol)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (ucontrol)
+		alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
+				    ucontrol->value.integer.value[0] ||
+				    ucontrol->value.integer.value[1]);
 }
 
 static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
@@ -3283,9 +3289,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
 	};
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook;
-		spec->gen.cap_sync_hook = alc269_fixup_hp_gpio_mic_mute_hook;
+		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
+		spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook;
 		spec->gpio_led = 0;
+		spec->mute_led_polarity = 0;
+		spec->gpio_mute_led_mask = 0x08;
+		spec->gpio_mic_led_mask = 0x10;
 		snd_hda_add_verbs(codec, gpio_init);
 	}
 }
@@ -3327,9 +3336,11 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
 	};
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook;
+		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
 		spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
 		spec->gpio_led = 0;
+		spec->mute_led_polarity = 0;
+		spec->gpio_mute_led_mask = 0x08;
 		spec->cap_mute_led_nid = 0x18;
 		snd_hda_add_verbs(codec, gpio_init);
 		codec->power_filter = led_power_filter;
@@ -3348,9 +3359,11 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec,
 	};
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook;
+		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
 		spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
 		spec->gpio_led = 0;
+		spec->mute_led_polarity = 0;
+		spec->gpio_mute_led_mask = 0x08;
 		spec->cap_mute_led_nid = 0x18;
 		snd_hda_add_verbs(codec, gpio_init);
 		codec->power_filter = led_power_filter;
@@ -5624,22 +5637,6 @@ static void alc_fixup_bass_chmap(struct hda_codec *codec,
 	}
 }
 
-/* turn on/off mute LED per vmaster hook */
-static void alc662_led_gpio1_mute_hook(void *private_data, int enabled)
-{
-	struct hda_codec *codec = private_data;
-	struct alc_spec *spec = codec->spec;
-	unsigned int oldval = spec->gpio_led;
-
-	if (enabled)
-		spec->gpio_led |= 0x01;
-	else
-		spec->gpio_led &= ~0x01;
-	if (spec->gpio_led != oldval)
-		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-				    spec->gpio_led);
-}
-
 /* avoid D3 for keeping GPIO up */
 static unsigned int gpio_led_power_filter(struct hda_codec *codec,
 					  hda_nid_t nid,
@@ -5662,8 +5659,10 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec,
 	};
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc662_led_gpio1_mute_hook;
+		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
 		spec->gpio_led = 0;
+		spec->mute_led_polarity = 1;
+		spec->gpio_mute_led_mask = 0x01;
 		snd_hda_add_verbs(codec, gpio_init);
 		codec->power_filter = gpio_led_power_filter;
 	}
-- 
cgit v1.2.3


From eaa8e5ef18fa9e09286482a4ded3a3cad36e44b2 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Fri, 21 Nov 2014 15:49:11 +0800
Subject: ALSA: hda/realtek - Supported HP mute Led for ALC286

New HP machine supported output mute led and input mute led.
ALC286:
GPIO1 to control output mute led.
GPIO5 to control input mute led.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++
 1 file changed, 27 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3c29a558e7db..3fcb7d951572 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3299,6 +3299,27 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
 	}
 }
 
+static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	static const struct hda_verb gpio_init[] = {
+		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x22 },
+		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x22 },
+		{}
+	};
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
+		spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook;
+		spec->gpio_led = 0;
+		spec->mute_led_polarity = 0;
+		spec->gpio_mute_led_mask = 0x02;
+		spec->gpio_mic_led_mask = 0x20;
+		snd_hda_add_verbs(codec, gpio_init);
+	}
+}
+
 /* turn on/off mic-mute LED per capture hook */
 static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec,
 					       struct snd_kcontrol *kcontrol,
@@ -4238,6 +4259,7 @@ enum {
 	ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
 	ALC282_FIXUP_ASPIRE_V5_PINS,
 	ALC280_FIXUP_HP_GPIO4,
+	ALC286_FIXUP_HP_GPIO_LED,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -4705,6 +4727,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc280_fixup_hp_gpio4,
 	},
+	[ALC286_FIXUP_HP_GPIO_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc286_fixup_hp_gpio_led,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -4745,6 +4771,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
-- 
cgit v1.2.3


From e9886ab06c1ef42451307c9367e344b2d8140e0b Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:48 +1300
Subject: ALSA: asihpi: Minor string and dead code cleanup

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 13 ++++---------
 1 file changed, 4 insertions(+), 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index ff9f9f1c0391..0e130dd8732e 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -28,7 +28,6 @@
 #include "hpioctl.h"
 #include "hpicmn.h"
 
-
 #include <linux/pci.h>
 #include <linux/init.h>
 #include <linux/jiffies.h>
@@ -47,7 +46,7 @@
 
 MODULE_LICENSE("GPL");
 MODULE_AUTHOR("AudioScience inc. <support@audioscience.com>");
-MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx "
+MODULE_DESCRIPTION("AudioScience ALSA ASI5xxx ASI6xxx ASI87xx ASI89xx "
 			HPI_VER_STRING);
 
 #if defined CONFIG_SND_DEBUG_VERBOSE
@@ -87,11 +86,11 @@ MODULE_PARM_DESC(enable_hpi_hwdep,
 #ifdef KERNEL_ALSA_BUILD
 static char *build_info = "Built using headers from kernel source";
 module_param(build_info, charp, S_IRUGO);
-MODULE_PARM_DESC(build_info, "built using headers from kernel source");
+MODULE_PARM_DESC(build_info, "Built using headers from kernel source");
 #else
 static char *build_info = "Built within ALSA source";
 module_param(build_info, charp, S_IRUGO);
-MODULE_PARM_DESC(build_info, "built within ALSA source");
+MODULE_PARM_DESC(build_info, "Built within ALSA source");
 #endif
 
 /* set to 1 to dump every control from adapter to log */
@@ -538,7 +537,7 @@ static void snd_card_asihpi_pcm_timer_start(struct snd_pcm_substream *
 	int expiry;
 
 	expiry = HZ / 200;
-	/*? (dpcm->period_bytes * HZ / dpcm->bytes_per_sec); */
+
 	expiry = max(expiry, 1); /* don't let it be zero! */
 	dpcm->timer.expires = jiffies + expiry;
 	dpcm->respawn_timer = 1;
@@ -2932,10 +2931,6 @@ static struct pci_driver driver = {
 	.id_table = asihpi_pci_tbl,
 	.probe = snd_asihpi_probe,
 	.remove = snd_asihpi_remove,
-#ifdef CONFIG_PM_SLEEP
-/*	.suspend = snd_asihpi_suspend,
-	.resume = snd_asihpi_resume, */
-#endif
 };
 
 static int __init snd_asihpi_init(void)
-- 
cgit v1.2.3


From 3872f19d96a55ec1d1e7af904d84457d91ef5a63 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:49 +1300
Subject: ALSA: asihpi: New I/O types - AVB & BLUlink, DAB Rf receiver

Audio cards wth have AVB or BLU Link IO.
Tuner card with DAB receiver

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 25 ++++++++++++++++++-------
 sound/pci/asihpi/hpi.h    | 16 ++++++++++++----
 2 files changed, 30 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 0e130dd8732e..628ef7f146d9 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -109,7 +109,7 @@ static int adapter_fs = DEFAULT_SAMPLERATE;
 struct clk_source {
 	int source;
 	int index;
-	char *name;
+	const char *name;
 };
 
 struct clk_cache {
@@ -1292,8 +1292,9 @@ static const char * const asihpi_tuner_band_names[] = {
 	"TV PAL I",
 	"TV PAL DK",
 	"TV SECAM",
+	"TV DAB",
 };
-
+/* Number of strings must match the enumerations for HPI_TUNER_BAND in hpi.h */
 compile_time_assert(
 	(ARRAY_SIZE(asihpi_tuner_band_names) ==
 		(HPI_TUNER_BAND_LAST+1)),
@@ -1313,9 +1314,11 @@ static const char * const asihpi_src_names[] = {
 	"Analog",
 	"Adapter",
 	"RTP",
-	"Internal"
+	"Internal",
+	"AVB",
+	"BLU-Link"
 };
-
+/* Number of strings must match the enumerations for HPI_SOURCENODES in hpi.h */
 compile_time_assert(
 	(ARRAY_SIZE(asihpi_src_names) ==
 		(HPI_SOURCENODE_LAST_INDEX-HPI_SOURCENODE_NONE+1)),
@@ -1331,8 +1334,11 @@ static const char * const asihpi_dst_names[] = {
 	"Net",
 	"Analog",
 	"RTP",
+	"AVB",
+	"Internal",
+	"BLU-Link"
 };
-
+/* Number of strings must match the enumerations for HPI_DESTNODES in hpi.h */
 compile_time_assert(
 	(ARRAY_SIZE(asihpi_dst_names) ==
 		(HPI_DESTNODE_LAST_INDEX-HPI_DESTNODE_NONE+1)),
@@ -2288,13 +2294,18 @@ static int snd_asihpi_cmode_add(struct snd_card_asihpi *asihpi,
 /*------------------------------------------------------------
    Sampleclock source  controls
  ------------------------------------------------------------*/
-static char *sampleclock_sources[MAX_CLOCKSOURCES] = {
+static const char const *sampleclock_sources[] = {
 	"N/A", "Local PLL", "Digital Sync", "Word External", "Word Header",
 	"SMPTE", "Digital1", "Auto", "Network", "Invalid",
-	"Prev Module",
+	"Prev Module", "BLU-Link",
 	"Digital2", "Digital3", "Digital4", "Digital5",
 	"Digital6", "Digital7", "Digital8"};
 
+	/* Number of strings must match expected enumerated values */
+	compile_time_assert(
+		(ARRAY_SIZE(sampleclock_sources) == MAX_CLOCKSOURCES),
+		assert_sampleclock_sources_size);
+
 static int snd_asihpi_clksrc_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 20887241a3ae..4466bd2c5272 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -196,8 +196,10 @@ enum HPI_SOURCENODES {
 	    packets of RTP audio samples from other devices. */
 	HPI_SOURCENODE_RTP_DESTINATION = 112,
 	HPI_SOURCENODE_INTERNAL = 113,	     /**< node internal to the device. */
+	HPI_SOURCENODE_AVB = 114,	     /**< AVB input stream */
+	HPI_SOURCENODE_BLULINK = 115,	     /**< BLU-link input channel */
 	/* !!!Update this  AND hpidebug.h if you add a new sourcenode type!!! */
-	HPI_SOURCENODE_LAST_INDEX = 113	     /**< largest ID */
+	HPI_SOURCENODE_LAST_INDEX = 115	     /**< largest ID */
 		/* AX6 max sourcenode types = 15 */
 };
 
@@ -224,8 +226,11 @@ enum HPI_DESTNODES {
 	/** RTP stream output node - This node is a source for
 	    packets of RTP audio samples that are sent to other devices. */
 	HPI_DESTNODE_RTP_SOURCE = 208,
+	HPI_DESTNODE_AVB = 209,		     /**< AVB output stream */
+	HPI_DESTNODE_INTERNAL = 210,	     /**< node internal to the device. */
+	HPI_DESTNODE_BLULINK = 211,	     /**< BLU-link output channel. */
 	/* !!!Update this AND hpidebug.h if you add a new destnode type!!! */
-	HPI_DESTNODE_LAST_INDEX = 208	     /**< largest ID */
+	HPI_DESTNODE_LAST_INDEX = 211	     /**< largest ID */
 		/* AX6 max destnode types = 15 */
 };
 
@@ -752,7 +757,8 @@ enum HPI_TUNER_BAND {
 	HPI_TUNER_BAND_TV_PAL_I = 7,	 /**< PAL-I TV band*/
 	HPI_TUNER_BAND_TV_PAL_DK = 8,	 /**< PAL-D/K TV band*/
 	HPI_TUNER_BAND_TV_SECAM_L = 9,	 /**< SECAM-L TV band*/
-	HPI_TUNER_BAND_LAST = 9	/**< the index of the last tuner band. */
+	HPI_TUNER_BAND_DAB = 10,
+	HPI_TUNER_BAND_LAST = 10 /**< the index of the last tuner band. */
 };
 
 /** Tuner mode attributes
@@ -842,8 +848,10 @@ enum HPI_SAMPLECLOCK_SOURCES {
 	HPI_SAMPLECLOCK_SOURCE_NETWORK = 8,
 /** From previous adjacent module (ASI2416 only)*/
 	HPI_SAMPLECLOCK_SOURCE_PREV_MODULE = 10,
+/** Blu link sample clock*/
+	HPI_SAMPLECLOCK_SOURCE_BLULINK = 11,
 /*! Update this if you add a new clock source.*/
-	HPI_SAMPLECLOCK_SOURCE_LAST = 10
+	HPI_SAMPLECLOCK_SOURCE_LAST = 11
 };
 
 /** Equalizer filter types. Used by HPI_ParametricEq_SetBand()
-- 
cgit v1.2.3


From 35a8dc1f66a0fa88144fcbcd562eb2b2c1e36f11 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:50 +1300
Subject: ALSA: asihpi: Logging format improvements

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 85 ++++++++++++++++++++++-------------------------
 1 file changed, 39 insertions(+), 46 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 628ef7f146d9..c06903304e12 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -288,21 +288,17 @@ static void print_hwparams(struct snd_pcm_substream *substream,
 {
 	char name[16];
 	snd_pcm_debug_name(substream, name, sizeof(name));
-	snd_printd("%s HWPARAMS\n", name);
-	snd_printd(" samplerate %d Hz\n", params_rate(p));
-	snd_printd(" channels %d\n", params_channels(p));
-	snd_printd(" format %d\n", params_format(p));
-	snd_printd(" subformat %d\n", params_subformat(p));
-	snd_printd(" buffer %d B\n", params_buffer_bytes(p));
-	snd_printd(" period %d B\n", params_period_bytes(p));
-	snd_printd(" access %d\n", params_access(p));
-	snd_printd(" period_size %d\n", params_period_size(p));
-	snd_printd(" periods %d\n", params_periods(p));
-	snd_printd(" buffer_size %d\n", params_buffer_size(p));
-	snd_printd(" %d B/s\n", params_rate(p) *
-		params_channels(p) *
+	snd_printdd("%s HWPARAMS\n", name);
+	snd_printdd(" samplerate=%dHz channels=%d format=%d subformat=%d\n",
+		params_rate(p), params_channels(p),
+		params_format(p), params_subformat(p));
+	snd_printdd(" buffer=%dB period=%dB period_size=%dB periods=%d\n",
+		params_buffer_bytes(p), params_period_bytes(p),
+		params_period_size(p), params_periods(p));
+	snd_printdd(" buffer_size=%d access=%d data_rate=%dB/s\n",
+		params_buffer_size(p), params_access(p),
+		params_rate(p) * params_channels(p) *
 		snd_pcm_format_width(params_format(p)) / 8);
-
 }
 
 static snd_pcm_format_t hpi_to_alsa_formats[] = {
@@ -480,7 +476,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 			params_buffer_bytes(params),  runtime->dma_addr);
 		if (err == 0) {
 			snd_printdd(
-				"stream_host_buffer_attach succeeded %u %lu\n",
+				"stream_host_buffer_attach success %u %lu\n",
 				params_buffer_bytes(params),
 				(unsigned long)runtime->dma_addr);
 		} else {
@@ -490,12 +486,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 
 		err = hpi_stream_get_info_ex(dpcm->h_stream, NULL,
-						&dpcm->hpi_buffer_attached,
-						NULL, NULL, NULL);
-
-		snd_printdd("stream_host_buffer_attach status 0x%x\n",
-				dpcm->hpi_buffer_attached);
-
+				&dpcm->hpi_buffer_attached, NULL, NULL, NULL);
 	}
 	bytes_per_sec = params_rate(params) * params_channels(params);
 	width = snd_pcm_format_width(params_format(params));
@@ -563,10 +554,10 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 	char name[16];
 
 	snd_pcm_debug_name(substream, name, sizeof(name));
-	snd_printdd("%s trigger\n", name);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
+		snd_printdd("%s trigger start\n", name);
 		snd_pcm_group_for_each_entry(s, substream) {
 			struct snd_pcm_runtime *runtime = s->runtime;
 			struct snd_card_asihpi_pcm *ds = runtime->private_data;
@@ -587,7 +578,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 				* data??
 				*/
 				unsigned int preload = ds->period_bytes * 1;
-				snd_printddd("%d preload x%x\n", s->number, preload);
+				snd_printddd("%d preload %d\n", s->number, preload);
 				hpi_handle_error(hpi_outstream_write_buf(
 						ds->h_stream,
 						&runtime->dma_area[0],
@@ -610,7 +601,6 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 			} else
 				break;
 		}
-		snd_printdd("start\n");
 		/* start the master stream */
 		snd_card_asihpi_pcm_timer_start(substream);
 		if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ||
@@ -619,6 +609,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
+		snd_printdd("%s trigger stop\n", name);
 		snd_card_asihpi_pcm_timer_stop(substream);
 		snd_pcm_group_for_each_entry(s, substream) {
 			if (snd_pcm_substream_chip(s) != card)
@@ -637,7 +628,6 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 			} else
 				break;
 		}
-		snd_printdd("stop\n");
 
 		/* _prepare and _hwparams reset the stream */
 		hpi_handle_error(hpi_stream_stop(dpcm->h_stream));
@@ -650,12 +640,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		snd_printdd("pause release\n");
+		snd_printdd("%s trigger pause release\n", name);
 		hpi_handle_error(hpi_stream_start(dpcm->h_stream));
 		snd_card_asihpi_pcm_timer_start(substream);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		snd_printdd("pause\n");
+		snd_printdd("%s trigger pause push\n", name);
 		snd_card_asihpi_pcm_timer_stop(substream);
 		hpi_handle_error(hpi_stream_stop(dpcm->h_stream));
 		break;
@@ -730,7 +720,6 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 
 	snd_pcm_debug_name(substream, name, sizeof(name));
 
-	snd_printdd("%s snd_card_asihpi_timer_function\n", name);
 
 	/* find minimum newdata and buffer pos in group */
 	snd_pcm_group_for_each_entry(s, substream) {
@@ -790,19 +779,20 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 				newdata);
 		}
 
-		snd_printdd("hw_ptr 0x%04lX, appl_ptr 0x%04lX\n",
+		snd_printddd("timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n",
+			name, s->number, state,
+			ds->pcm_buf_elapsed_dma_ofs,
+			ds->pcm_buf_host_rw_ofs,
+			pcm_buf_dma_ofs,
+			(int)bytes_avail,
+
+			(int)on_card_bytes,
+			buffer_size-bytes_avail,
 			(unsigned long)frames_to_bytes(runtime,
 						runtime->status->hw_ptr),
 			(unsigned long)frames_to_bytes(runtime,
-						runtime->control->appl_ptr));
-
-		snd_printdd("%d S=%d, "
-			"rw=0x%04X, dma=0x%04X, left=0x%04X, "
-			"aux=0x%04X space=0x%04X\n",
-			s->number, state,
-			ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs,
-			(int)bytes_avail,
-			(int)on_card_bytes, buffer_size-bytes_avail);
+						runtime->control->appl_ptr)
+		);
 		loops++;
 	}
 	pcm_buf_dma_ofs = min_buf_pos;
@@ -820,7 +810,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 
 	next_jiffies = max(next_jiffies, 1U);
 	dpcm->timer.expires = jiffies + next_jiffies;
-	snd_printdd("jif %d buf pos 0x%04X newdata 0x%04X xfer 0x%04X\n",
+	snd_printddd("timer2, jif=%d, buf_pos=%d, newdata=%d, xfer=%d\n",
 			next_jiffies, pcm_buf_dma_ofs, newdata, xfercount);
 
 	snd_pcm_group_for_each_entry(s, substream) {
@@ -852,7 +842,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 			}
 
 			if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-				snd_printddd("P%d write1 0x%04X 0x%04X\n",
+				snd_printddd("write1, P=%d, xfer=%d, buf_ofs=%d\n",
 					s->number, xfer1, buf_ofs);
 				hpi_handle_error(
 					hpi_outstream_write_buf(
@@ -862,7 +852,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 				if (xfer2) {
 					pd = s->runtime->dma_area;
 
-					snd_printddd("P%d write2 0x%04X 0x%04X\n",
+					snd_printddd("write2, P=%d, xfer=%d, buf_ofs=%d\n",
 							s->number,
 							xfercount - xfer1, buf_ofs);
 					hpi_handle_error(
@@ -872,7 +862,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 							&ds->format));
 				}
 			} else {
-				snd_printddd("C%d read1 0x%04x\n",
+				snd_printddd("read1, C=%d, xfer=%d\n",
 					s->number, xfer1);
 				hpi_handle_error(
 					hpi_instream_read_buf(
@@ -880,7 +870,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 						pd, xfer1));
 				if (xfer2) {
 					pd = s->runtime->dma_area;
-					snd_printddd("C%d read2 0x%04x\n",
+					snd_printddd("read2, C=%d, xfer=%d\n",
 						s->number, xfer2);
 					hpi_handle_error(
 						hpi_instream_read_buf(
@@ -933,7 +923,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream)
 	snd_pcm_debug_name(substream, name, sizeof(name));
 
 	ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs  % dpcm->buffer_bytes);
-	snd_printddd("%s pointer = 0x%04lx\n", name, (unsigned long)ptr);
+	snd_printddd("%s, pointer=%ld\n", name, (unsigned long)ptr);
 	return ptr;
 }
 
@@ -1081,9 +1071,10 @@ snd_card_asihpi_capture_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
+	char name[16];
+	snd_pcm_debug_name(substream, name, sizeof(name));
 
-	snd_printddd("capture pointer %d=%d\n",
-			substream->number, dpcm->pcm_buf_dma_ofs);
+	snd_printddd("%s, pointer=%d\n", name, dpcm->pcm_buf_dma_ofs);
 	/* NOTE Unlike playback can't use actual samples_played
 		for the capture position, because those samples aren't yet in
 		the local buffer available for reading.
@@ -2867,6 +2858,8 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 		asihpi->in_min_chans = 1;
 	}
 
+	snd_printk(KERN_INFO "update_interval_frames: %d",
+		asihpi->update_interval_frames);
 	snd_printk(KERN_INFO "Has dma:%d, grouping:%d, mrx:%d\n",
 			asihpi->can_dma,
 			asihpi->support_grouping,
-- 
cgit v1.2.3


From e51c58c982a0f81baa6fdee7331a8700c8586be5 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:51 +1300
Subject: ALSA: asihpi: Use CONFIG_64BIT directly

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpi_internal.h | 16 ++++++++++------
 sound/pci/asihpi/hpios.h        |  4 ----
 2 files changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index bc86cb726d79..c9bdc284cdaf 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -554,17 +554,21 @@ struct hpi_pci {
 	struct pci_dev *pci_dev;
 };
 
+/** Adapter specification resource */
+struct hpi_adapter_specification {
+	u32 type;
+	u8 modules[4];
+};
+
 struct hpi_resource {
 	union {
 		const struct hpi_pci *pci;
 		const char *net_if;
+		struct hpi_adapter_specification adapter_spec;
+		const void *sw_if;
 	} r;
-#ifndef HPI64BIT		/* keep structure size constant */
-	u32 pad_to64;
-#endif
 	u16 bus_type;		/* HPI_BUS_PNPISA, _PCI, _USB etc */
 	u16 padding;
-
 };
 
 /** Format info used inside struct hpi_message
@@ -582,7 +586,7 @@ struct hpi_msg_format {
 struct hpi_msg_data {
 	struct hpi_msg_format format;
 	u8 *pb_data;
-#ifndef HPI64BIT
+#ifndef CONFIG_64BIT
 	u32 padding;
 #endif
 	u32 data_size;
@@ -595,7 +599,7 @@ struct hpi_data_legacy32 {
 	u32 data_size;
 };
 
-#ifdef HPI64BIT
+#ifdef CONFIG_64BIT
 /* Compatibility version of struct hpi_data*/
 struct hpi_data_compat32 {
 	struct hpi_msg_format format;
diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index d3fbd0d76c37..d17d017940d8 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -41,10 +41,6 @@ HPI Operating System Specific macros for Linux Kernel driver
 
 #define HPI_NO_OS_FILE_OPS
 
-#ifdef CONFIG_64BIT
-#define HPI64BIT
-#endif
-
 /** Details of a memory area allocated with  pci_alloc_consistent
 Need all info for parameters to pci_free_consistent
 */
-- 
cgit v1.2.3


From c1464a885444dd7e9c4491177ee102b64adc46c5 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:52 +1300
Subject: ALSA: asihpi: Refactor control cache code.

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpicmn.c | 107 +++++++++++++++++++++++++++-------------------
 1 file changed, 63 insertions(+), 44 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 7ed5c26c3737..c7751243dc42 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -206,6 +206,14 @@ static unsigned int control_cache_alloc_check(struct hpi_control_cache *pC)
 			struct hpi_control_cache_info *info =
 				(struct hpi_control_cache_info *)
 				&p_master_cache[byte_count];
+			u16 control_index = info->control_index;
+
+			if (control_index >= pC->control_count) {
+				HPI_DEBUG_LOG(INFO,
+					"adap %d control index %d out of range, cache not ready?\n",
+					pC->adap_idx, control_index);
+				return 0;
+			}
 
 			if (!info->size_in32bit_words) {
 				if (!i) {
@@ -225,10 +233,10 @@ static unsigned int control_cache_alloc_check(struct hpi_control_cache *pC)
 			}
 
 			if (info->control_type) {
-				pC->p_info[info->control_index] = info;
+				pC->p_info[control_index] = info;
 				cached++;
 			} else {	/* dummy cache entry */
-				pC->p_info[info->control_index] = NULL;
+				pC->p_info[control_index] = NULL;
 			}
 
 			byte_count += info->size_in32bit_words * 4;
@@ -309,35 +317,18 @@ static const struct pad_ofs_size pad_desc[] = {
 /** CheckControlCache checks the cache and fills the struct hpi_response
  * accordingly. It returns one if a cache hit occurred, zero otherwise.
  */
-short hpi_check_control_cache(struct hpi_control_cache *p_cache,
+short hpi_check_control_cache_single(struct hpi_control_cache_single *pC,
 	struct hpi_message *phm, struct hpi_response *phr)
 {
-	short found = 1;
-	struct hpi_control_cache_info *pI;
-	struct hpi_control_cache_single *pC;
 	size_t response_size;
-	if (!find_control(phm->obj_index, p_cache, &pI)) {
-		HPI_DEBUG_LOG(VERBOSE,
-			"HPICMN find_control() failed for adap %d\n",
-			phm->adapter_index);
-		return 0;
-	}
-
-	phr->error = 0;
-	phr->specific_error = 0;
-	phr->version = 0;
+	short found = 1;
 
 	/* set the default response size */
 	response_size =
 		sizeof(struct hpi_response_header) +
 		sizeof(struct hpi_control_res);
 
-	/* pC is the default cached control strucure. May be cast to
-	   something else in the following switch statement.
-	 */
-	pC = (struct hpi_control_cache_single *)pI;
-
-	switch (pI->control_type) {
+	switch (pC->u.i.control_type) {
 
 	case HPI_CONTROL_METER:
 		if (phm->u.c.attribute == HPI_METER_PEAK) {
@@ -467,7 +458,7 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 		break;
 	case HPI_CONTROL_PAD:{
 			struct hpi_control_cache_pad *p_pad;
-			p_pad = (struct hpi_control_cache_pad *)pI;
+			p_pad = (struct hpi_control_cache_pad *)pC;
 
 			if (!(p_pad->field_valid_flags & (1 <<
 						HPI_CTL_ATTR_INDEX(phm->u.c.
@@ -531,7 +522,8 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 
 	HPI_DEBUG_LOG(VERBOSE, "%s Adap %d, Ctl %d, Type %d, Attr %d\n",
 		found ? "Cached" : "Uncached", phm->adapter_index,
-		pI->control_index, pI->control_type, phm->u.c.attribute);
+		pC->u.i.control_index, pC->u.i.control_type,
+		phm->u.c.attribute);
 
 	if (found) {
 		phr->size = (u16)response_size;
@@ -543,34 +535,36 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 	return found;
 }
 
-/** Updates the cache with Set values.
-
-Only update if no error.
-Volume and Level return the limited values in the response, so use these
-Multiplexer does so use sent values
-*/
-void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache,
+short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 	struct hpi_message *phm, struct hpi_response *phr)
 {
-	struct hpi_control_cache_single *pC;
 	struct hpi_control_cache_info *pI;
 
-	if (phr->error)
-		return;
-
 	if (!find_control(phm->obj_index, p_cache, &pI)) {
 		HPI_DEBUG_LOG(VERBOSE,
 			"HPICMN find_control() failed for adap %d\n",
 			phm->adapter_index);
-		return;
+		return 0;
 	}
 
-	/* pC is the default cached control strucure.
-	   May be cast to something else in the following switch statement.
-	 */
-	pC = (struct hpi_control_cache_single *)pI;
+	phr->error = 0;
+	phr->specific_error = 0;
+	phr->version = 0;
+
+	return hpi_check_control_cache_single((struct hpi_control_cache_single
+			*)pI, phm, phr);
+}
+
+/** Updates the cache with Set values.
 
-	switch (pI->control_type) {
+Only update if no error.
+Volume and Level return the limited values in the response, so use these
+Multiplexer does so use sent values
+*/
+void hpi_cmn_control_cache_sync_to_msg_single(struct hpi_control_cache_single
+	*pC, struct hpi_message *phm, struct hpi_response *phr)
+{
+	switch (pC->u.i.control_type) {
 	case HPI_CONTROL_VOLUME:
 		if (phm->u.c.attribute == HPI_VOLUME_GAIN) {
 			pC->u.vol.an_log[0] = phr->u.c.an_log_value[0];
@@ -625,6 +619,30 @@ void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache,
 	}
 }
 
+void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache,
+	struct hpi_message *phm, struct hpi_response *phr)
+{
+	struct hpi_control_cache_single *pC;
+	struct hpi_control_cache_info *pI;
+
+	if (phr->error)
+		return;
+
+	if (!find_control(phm->obj_index, p_cache, &pI)) {
+		HPI_DEBUG_LOG(VERBOSE,
+			"HPICMN find_control() failed for adap %d\n",
+			phm->adapter_index);
+		return;
+	}
+
+	/* pC is the default cached control strucure.
+	   May be cast to something else in the following switch statement.
+	 */
+	pC = (struct hpi_control_cache_single *)pI;
+
+	hpi_cmn_control_cache_sync_to_msg_single(pC, phm, phr);
+}
+
 /** Allocate control cache.
 
 \return Cache pointer, or NULL if allocation fails.
@@ -637,12 +655,13 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
 	if (!p_cache)
 		return NULL;
 
-	p_cache->p_info = kcalloc(control_count, sizeof(*p_cache->p_info),
-				  GFP_KERNEL);
+	p_cache->p_info =
+		kcalloc(control_count, sizeof(*p_cache->p_info), GFP_KERNEL);
 	if (!p_cache->p_info) {
 		kfree(p_cache);
 		return NULL;
 	}
+
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = control_count;
 	p_cache->p_cache = p_dsp_control_buffer;
-- 
cgit v1.2.3


From f9a376c3f6d77e59d41350901b2bafbaf8791df0 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:53 +1300
Subject: ALSA: asihpi: Add support for stream interrupt.

Some cards have a so-called low-latency mode, in which they present
a single multichannel stream with no mixing or samplerate conversion.
In this mode the card can generate an interrupt per internal processing
block (typically 32 or 64 frames)

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c       | 177 +++++++++++++++++++++++++++++++++-------
 sound/pci/asihpi/hpi6205.c      |  43 ++++++++--
 sound/pci/asihpi/hpi_internal.h |   4 +-
 sound/pci/asihpi/hpicmn.h       |  19 ++++-
 sound/pci/asihpi/hpioctl.c      | 124 ++++++++++++++++++++++++++--
 sound/pci/asihpi/hpios.h        |   4 +
 6 files changed, 321 insertions(+), 50 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index c06903304e12..ae29f30547cc 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -1,6 +1,6 @@
 /*
  *  Asihpi soundcard
- *  Copyright (c) by AudioScience Inc <alsa@audioscience.com>
+ *  Copyright (c) by AudioScience Inc <support@audioscience.com>
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of version 2 of the GNU General Public License as
@@ -124,6 +124,16 @@ struct snd_card_asihpi {
 	struct pci_dev *pci;
 	struct hpi_adapter *hpi;
 
+	/* In low latency mode there is only one stream, a pointer to its
+	 * private data is stored here on trigger and cleared on stop.
+	 * The interrupt handler uses it as a parameter when calling
+	 * snd_card_asihpi_timer_function().
+	 */
+	struct snd_card_asihpi_pcm *llmode_streampriv;
+	struct tasklet_struct t;
+	void (*pcm_start)(struct snd_pcm_substream *substream);
+	void (*pcm_stop)(struct snd_pcm_substream *substream);
+
 	u32 h_mixer;
 	struct clk_cache cc;
 
@@ -544,6 +554,48 @@ static void snd_card_asihpi_pcm_timer_stop(struct snd_pcm_substream *substream)
 	del_timer(&dpcm->timer);
 }
 
+static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream)
+{
+	struct snd_card_asihpi_pcm *dpcm;
+	struct snd_card_asihpi *card;
+
+	BUG_ON(!substream);
+
+	dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data;
+	card = snd_pcm_substream_chip(substream);
+
+	BUG_ON(in_interrupt());
+	tasklet_disable(&card->t);
+	card->llmode_streampriv = dpcm;
+	tasklet_enable(&card->t);
+
+	hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index,
+		HPI_ADAPTER_PROPERTY_IRQ_RATE,
+		card->update_interval_frames, 0));
+}
+
+static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_card_asihpi_pcm *dpcm;
+	struct snd_card_asihpi *card;
+
+	BUG_ON(!substream);
+
+	dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data;
+	card = snd_pcm_substream_chip(substream);
+
+	hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index,
+		HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0));
+
+	if (in_interrupt())
+		card->llmode_streampriv = NULL;
+	else {
+		tasklet_disable(&card->t);
+		card->llmode_streampriv = NULL;
+		tasklet_enable(&card->t);
+	}
+}
+
 static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 					   int cmd)
 {
@@ -602,7 +654,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 				break;
 		}
 		/* start the master stream */
-		snd_card_asihpi_pcm_timer_start(substream);
+		card->pcm_start(substream);
 		if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ||
 			!card->can_dma)
 			hpi_handle_error(hpi_stream_start(dpcm->h_stream));
@@ -610,7 +662,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 
 	case SNDRV_PCM_TRIGGER_STOP:
 		snd_printdd("%s trigger stop\n", name);
-		snd_card_asihpi_pcm_timer_stop(substream);
+		card->pcm_stop(substream);
 		snd_pcm_group_for_each_entry(s, substream) {
 			if (snd_pcm_substream_chip(s) != card)
 				continue;
@@ -641,12 +693,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		snd_printdd("%s trigger pause release\n", name);
+		card->pcm_start(substream);
 		hpi_handle_error(hpi_stream_start(dpcm->h_stream));
-		snd_card_asihpi_pcm_timer_start(substream);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		snd_printdd("%s trigger pause push\n", name);
-		snd_card_asihpi_pcm_timer_stop(substream);
+		card->pcm_stop(substream);
 		hpi_handle_error(hpi_stream_stop(dpcm->h_stream));
 		break;
 	default:
@@ -718,8 +770,8 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 	u32 buffer_size, bytes_avail, samples_played, on_card_bytes;
 	char name[16];
 
-	snd_pcm_debug_name(substream, name, sizeof(name));
 
+	snd_pcm_debug_name(substream, name, sizeof(name));
 
 	/* find minimum newdata and buffer pos in group */
 	snd_pcm_group_for_each_entry(s, substream) {
@@ -779,7 +831,8 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 				newdata);
 		}
 
-		snd_printddd("timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n",
+		snd_printddd(
+			"timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n",
 			name, s->number, state,
 			ds->pcm_buf_elapsed_dma_ofs,
 			ds->pcm_buf_host_rw_ofs,
@@ -815,11 +868,13 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 
 	snd_pcm_group_for_each_entry(s, substream) {
 		struct snd_card_asihpi_pcm *ds = s->runtime->private_data;
+		runtime = s->runtime;
 
 		/* don't link Cap and Play */
 		if (substream->stream != s->stream)
 			continue;
 
+		/* Store dma offset for use by pointer callback */
 		ds->pcm_buf_dma_ofs = pcm_buf_dma_ofs;
 
 		if (xfercount &&
@@ -878,16 +933,38 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 							pd, xfer2));
 				}
 			}
+			/* ? host_rw_ofs always ahead of elapsed_dma_ofs by preload size? */
 			ds->pcm_buf_host_rw_ofs += xfercount;
 			ds->pcm_buf_elapsed_dma_ofs += xfercount;
 			snd_pcm_period_elapsed(s);
 		}
 	}
 
-	if (dpcm->respawn_timer)
+	if (!card->hpi->interrupt_mode && dpcm->respawn_timer)
 		add_timer(&dpcm->timer);
 }
 
+static void snd_card_asihpi_int_task(unsigned long data)
+{
+	struct hpi_adapter *a = (struct hpi_adapter *)data;
+	struct snd_card_asihpi *asihpi;
+
+	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
+	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
+	if (asihpi->llmode_streampriv)
+		snd_card_asihpi_timer_function(
+			(unsigned long)asihpi->llmode_streampriv);
+}
+
+static void snd_card_asihpi_isr(struct hpi_adapter *a)
+{
+	struct snd_card_asihpi *asihpi;
+
+	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
+	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
+	tasklet_schedule(&asihpi->t);
+}
+
 /***************************** PLAYBACK OPS ****************/
 static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream,
 					  unsigned int cmd, void *arg)
@@ -995,13 +1072,22 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
 	runtime->private_free = snd_card_asihpi_runtime_free;
 
 	memset(&snd_card_asihpi_playback, 0, sizeof(snd_card_asihpi_playback));
-	snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX;
-	snd_card_asihpi_playback.period_bytes_min = PERIOD_BYTES_MIN;
-	/*?snd_card_asihpi_playback.period_bytes_min =
-	card->out_max_chans * 4096; */
-	snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN;
-	snd_card_asihpi_playback.periods_min = PERIODS_MIN;
-	snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN;
+	if (!card->hpi->interrupt_mode) {
+		snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX;
+		snd_card_asihpi_playback.period_bytes_min = PERIOD_BYTES_MIN;
+		snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN;
+		snd_card_asihpi_playback.periods_min = PERIODS_MIN;
+		snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN;
+	} else {
+		size_t pbmin = card->update_interval_frames *
+			card->out_max_chans;
+		snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX;
+		snd_card_asihpi_playback.period_bytes_min = pbmin;
+		snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN;
+		snd_card_asihpi_playback.periods_min = PERIODS_MIN;
+		snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / pbmin;
+	}
+
 	/* snd_card_asihpi_playback.fifo_size = 0; */
 	snd_card_asihpi_playback.channels_max = card->out_max_chans;
 	snd_card_asihpi_playback.channels_min = card->out_min_chans;
@@ -1036,7 +1122,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
 		card->update_interval_frames);
 
 	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-		card->update_interval_frames * 2, UINT_MAX);
+		card->update_interval_frames, UINT_MAX);
 
 	snd_printdd("playback open\n");
 
@@ -1102,8 +1188,6 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-
-
 static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi,
 					u32 h_stream)
 {
@@ -1170,11 +1254,21 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream)
 	runtime->private_free = snd_card_asihpi_runtime_free;
 
 	memset(&snd_card_asihpi_capture, 0, sizeof(snd_card_asihpi_capture));
-	snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX;
-	snd_card_asihpi_capture.period_bytes_min = PERIOD_BYTES_MIN;
-	snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN;
-	snd_card_asihpi_capture.periods_min = PERIODS_MIN;
-	snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN;
+	if (!card->hpi->interrupt_mode) {
+		snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX;
+		snd_card_asihpi_capture.period_bytes_min = PERIOD_BYTES_MIN;
+		snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN;
+		snd_card_asihpi_capture.periods_min = PERIODS_MIN;
+		snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN;
+	} else {
+		size_t pbmin = card->update_interval_frames *
+			card->out_max_chans;
+		snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX;
+		snd_card_asihpi_capture.period_bytes_min = pbmin;
+		snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN;
+		snd_card_asihpi_capture.periods_min = PERIODS_MIN;
+		snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / pbmin;
+	}
 	/* snd_card_asihpi_capture.fifo_size = 0; */
 	snd_card_asihpi_capture.channels_max = card->in_max_chans;
 	snd_card_asihpi_capture.channels_min = card->in_min_chans;
@@ -1199,7 +1293,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
 		card->update_interval_frames);
 	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-		card->update_interval_frames * 2, UINT_MAX);
+		card->update_interval_frames, UINT_MAX);
 
 	snd_pcm_set_sync(substream);
 
@@ -2444,15 +2538,19 @@ static int snd_asihpi_clkrate_get(struct snd_kcontrol *kcontrol,
 static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi,
 				      struct hpi_control *hpi_ctl)
 {
-	struct snd_card *card = asihpi->card;
+	struct snd_card *card;
 	struct snd_kcontrol_new snd_control;
 
-	struct clk_cache *clkcache = &asihpi->cc;
+	struct clk_cache *clkcache;
 	u32 hSC =  hpi_ctl->h_control;
 	int has_aes_in = 0;
 	int i, j;
 	u16 source;
 
+	if (snd_BUG_ON(!asihpi))
+		return -EINVAL;
+	card = asihpi->card;
+	clkcache = &asihpi->cc;
 	snd_control.private_value = hpi_ctl->h_control;
 
 	clkcache->has_local = 0;
@@ -2808,6 +2906,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	asihpi->card = card;
 	asihpi->pci = pci_dev;
 	asihpi->hpi = hpi;
+	hpi->snd_card = card;
 
 	snd_printk(KERN_INFO "adapter ID=%4X index=%d\n",
 			asihpi->hpi->adapter->type, adapter_index);
@@ -2830,8 +2929,16 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	if (err)
 		asihpi->update_interval_frames = 512;
 
-	if (!asihpi->can_dma)
-		asihpi->update_interval_frames *= 2;
+	if (hpi->interrupt_mode) {
+		asihpi->pcm_start = snd_card_asihpi_pcm_int_start;
+		asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop;
+		tasklet_init(&asihpi->t, snd_card_asihpi_int_task,
+			(unsigned long)hpi);
+		hpi->interrupt_callback = snd_card_asihpi_isr;
+	} else {
+		asihpi->pcm_start = snd_card_asihpi_pcm_timer_start;
+		asihpi->pcm_stop = snd_card_asihpi_pcm_timer_stop;
+	}
 
 	hpi_handle_error(hpi_instream_open(adapter_index,
 			     0, &h_stream));
@@ -2841,6 +2948,9 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 
 	hpi_handle_error(hpi_instream_close(h_stream));
 
+	if (!asihpi->can_dma)
+		asihpi->update_interval_frames *= 2;
+
 	err = hpi_adapter_get_property(adapter_index,
 		HPI_ADAPTER_PROPERTY_CURCHANNELS,
 		&asihpi->in_max_chans, &asihpi->out_max_chans);
@@ -2900,7 +3010,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	err = snd_card_register(card);
 
 	if (!err) {
-		hpi->snd_card = card;
 		dev++;
 		return 0;
 	}
@@ -2914,6 +3023,16 @@ __nodev:
 static void snd_asihpi_remove(struct pci_dev *pci_dev)
 {
 	struct hpi_adapter *hpi = pci_get_drvdata(pci_dev);
+	struct snd_card_asihpi *asihpi = hpi->snd_card->private_data;
+
+	/* Stop interrupts */
+	if (hpi->interrupt_mode) {
+		hpi->interrupt_callback = NULL;
+		hpi_handle_error(hpi_adapter_set_property(hpi->adapter->index,
+			HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0));
+		tasklet_kill(&asihpi->t);
+	}
+
 	snd_card_free(hpi->snd_card);
 	hpi->snd_card = NULL;
 	asihpi_adapter_remove(pci_dev);
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 4f2873880b16..8d5abfa4e24b 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -163,6 +163,9 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 
 static void delete_adapter_obj(struct hpi_adapter_obj *pao);
 
+static int adapter_irq_query_and_clear(struct hpi_adapter_obj *pao,
+	u32 message);
+
 static void outstream_host_buffer_allocate(struct hpi_adapter_obj *pao,
 	struct hpi_message *phm, struct hpi_response *phr);
 
@@ -283,7 +286,6 @@ static void adapter_message(struct hpi_adapter_obj *pao,
 	case HPI_ADAPTER_DELETE:
 		adapter_delete(pao, phm, phr);
 		break;
-
 	default:
 		hw_message(pao, phm, phr);
 		break;
@@ -673,6 +675,12 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 
 	HPI_DEBUG_LOG(INFO, "bootload DSP OK\n");
 
+	pao->irq_query_and_clear = adapter_irq_query_and_clear;
+	pao->instream_host_buffer_status =
+		phw->p_interface_buffer->instream_host_buffer_status;
+	pao->outstream_host_buffer_status =
+		phw->p_interface_buffer->outstream_host_buffer_status;
+
 	return hpi_add_adapter(pao);
 }
 
@@ -713,6 +721,21 @@ static void delete_adapter_obj(struct hpi_adapter_obj *pao)
 
 /*****************************************************************************/
 /* Adapter functions */
+static int adapter_irq_query_and_clear(struct hpi_adapter_obj *pao,
+	u32 message)
+{
+	struct hpi_hw_obj *phw = pao->priv;
+	u32 hsr = 0;
+
+	hsr = ioread32(phw->prHSR);
+	if (hsr & C6205_HSR_INTSRC) {
+		/* reset the interrupt from the DSP */
+		iowrite32(C6205_HSR_INTSRC, phw->prHSR);
+		return HPI_IRQ_MIXER;
+	}
+
+	return HPI_IRQ_NONE;
+}
 
 /*****************************************************************************/
 /* OutStream Host buffer functions */
@@ -1331,17 +1354,21 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 	if (boot_code_id[1] != 0) {
 		/* DSP 1 is a C6713 */
 		/* CLKX0 <- '1' release the C6205 bootmode pulldowns */
-		boot_loader_write_mem32(pao, 0, (0x018C0024L), 0x00002202);
+		boot_loader_write_mem32(pao, 0, 0x018C0024, 0x00002202);
 		hpios_delay_micro_seconds(100);
 		/* Reset the 6713 #1 - revB */
 		boot_loader_write_mem32(pao, 0, C6205_BAR0_TIMER1_CTL, 0);
-
-		/* dummy read every 4 words for 6205 advisory 1.4.4 */
-		boot_loader_read_mem32(pao, 0, 0);
-
+		/* value of bit 3 is unknown after DSP reset, other bits shoudl be 0 */
+		if (0 != (boot_loader_read_mem32(pao, 0,
+					(C6205_BAR0_TIMER1_CTL)) & ~8))
+			return HPI6205_ERROR_6205_REG;
 		hpios_delay_micro_seconds(100);
+
 		/* Release C6713 from reset - revB */
 		boot_loader_write_mem32(pao, 0, C6205_BAR0_TIMER1_CTL, 4);
+		if (4 != (boot_loader_read_mem32(pao, 0,
+					(C6205_BAR0_TIMER1_CTL)) & ~8))
+			return HPI6205_ERROR_6205_REG;
 		hpios_delay_micro_seconds(100);
 	}
 
@@ -2089,7 +2116,7 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao,
 		return 0;
 	}
 
-	/* Assume buffer of type struct bus_master_interface
+	/* Assume buffer of type struct bus_master_interface_62
 	   is allocated "noncacheable" */
 
 	if (!wait_dsp_ack(phw, H620_HIF_IDLE, HPI6205_TIMEOUT)) {
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index c9bdc284cdaf..48380ce2c81b 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -686,8 +686,8 @@ union hpi_adapterx_msg {
 		u16 value;
 	} test_assert;
 	struct {
-		u32 yes;
-	} irq_query;
+		u32 message;
+	} irq;
 	u32 pad[3];
 };
 
diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h
index e44121283047..46629c2d101b 100644
--- a/sound/pci/asihpi/hpicmn.h
+++ b/sound/pci/asihpi/hpicmn.h
@@ -1,7 +1,7 @@
 /**
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -21,7 +21,11 @@
 struct hpi_adapter_obj;
 
 /* a function that takes an adapter obj and returns an int */
-typedef int adapter_int_func(struct hpi_adapter_obj *pao);
+typedef int adapter_int_func(struct hpi_adapter_obj *pao, u32 message);
+
+#define HPI_IRQ_NONE		(0)
+#define HPI_IRQ_MESSAGE		(1)
+#define HPI_IRQ_MIXER		(2)
 
 struct hpi_adapter_obj {
 	struct hpi_pci pci;	/* PCI info - bus#,dev#,address etc */
@@ -33,6 +37,9 @@ struct hpi_adapter_obj {
 	u16 dsp_crashed;
 	u16 has_control_cache;
 	void *priv;
+	adapter_int_func *irq_query_and_clear;
+	struct hpi_hostbuffer_status *instream_host_buffer_status;
+	struct hpi_hostbuffer_status *outstream_host_buffer_status;
 };
 
 struct hpi_control_cache {
@@ -55,13 +62,21 @@ void hpi_delete_adapter(struct hpi_adapter_obj *pao);
 
 short hpi_check_control_cache(struct hpi_control_cache *pC,
 	struct hpi_message *phm, struct hpi_response *phr);
+
+short hpi_check_control_cache_single(struct hpi_control_cache_single *pC,
+	struct hpi_message *phm, struct hpi_response *phr);
+
 struct hpi_control_cache *hpi_alloc_control_cache(const u32
 	number_of_controls, const u32 size_in_bytes, u8 *pDSP_control_buffer);
+
 void hpi_free_control_cache(struct hpi_control_cache *p_cache);
 
 void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *pC,
 	struct hpi_message *phm, struct hpi_response *phr);
 
+void hpi_cmn_control_cache_sync_to_msg_single(struct hpi_control_cache_single
+	*pC, struct hpi_message *phm, struct hpi_response *phr);
+
 u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr);
 
 hpi_handler_func HPI_COMMON;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 7f0272032fbb..9454932fc9c0 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -1,7 +1,8 @@
 /*******************************************************************************
-
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Common Linux HPI ioctl and module probe/remove functions
+
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -12,11 +13,6 @@
     MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
     GNU General Public License for more details.
 
-    You should have received a copy of the GNU General Public License
-    along with this program; if not, write to the Free Software
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-Common Linux HPI ioctl and module probe/remove functions
 *******************************************************************************/
 #define SOURCEFILE_NAME "hpioctl.c"
 
@@ -29,6 +25,7 @@ Common Linux HPI ioctl and module probe/remove functions
 #include "hpicmn.h"
 
 #include <linux/fs.h>
+#include <linux/interrupt.h>
 #include <linux/slab.h>
 #include <linux/moduleparam.h>
 #include <asm/uaccess.h>
@@ -307,10 +304,38 @@ out:
 	return err;
 }
 
+static int asihpi_irq_count;
+
+static irqreturn_t asihpi_isr(int irq, void *dev_id)
+{
+	struct hpi_adapter *a = dev_id;
+	int handled;
+
+	if (!a->adapter->irq_query_and_clear) {
+		pr_err("asihpi_isr ASI%04X:%d no handler\n", a->adapter->type,
+			a->adapter->index);
+		return IRQ_NONE;
+	}
+
+	handled = a->adapter->irq_query_and_clear(a->adapter, 0);
+
+	if (!handled)
+		return IRQ_NONE;
+
+	asihpi_irq_count++;
+	/* printk(KERN_INFO "asihpi_isr %d ASI%04X:%d irq handled\n",
+	   asihpi_irq_count, a->adapter->type, a->adapter->index); */
+
+	if (a->interrupt_callback)
+		a->interrupt_callback(a);
+
+	return IRQ_HANDLED;
+}
+
 int asihpi_adapter_probe(struct pci_dev *pci_dev,
 			 const struct pci_device_id *pci_id)
 {
-	int idx, nm;
+	int idx, nm, low_latency_mode = 0, irq_supported = 0;
 	int adapter_index;
 	unsigned int memlen;
 	struct hpi_message hm;
@@ -388,8 +413,39 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	hm.adapter_index = adapter.adapter->index;
 	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
 
-	if (hr.error)
+	if (hr.error) {
+		HPI_DEBUG_LOG(ERROR, "HPI_ADAPTER_OPEN failed, aborting\n");
+		goto err;
+	}
+
+	/* Check if current mode == Low Latency mode */
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+		HPI_ADAPTER_GET_MODE);
+	hm.adapter_index = adapter.adapter->index;
+	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
+
+	if (hr.error) {
+		HPI_DEBUG_LOG(ERROR,
+			"HPI_ADAPTER_GET_MODE failed, aborting\n");
 		goto err;
+	}
+
+	if (hr.u.ax.mode.adapter_mode == HPI_ADAPTER_MODE_LOW_LATENCY)
+		low_latency_mode = 1;
+
+	/* Check if IRQs are supported */
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+		HPI_ADAPTER_GET_PROPERTY);
+	hm.adapter_index = adapter.adapter->index;
+	hm.u.ax.property_set.property = HPI_ADAPTER_PROPERTY_SUPPORTS_IRQ;
+	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
+	if (hr.error || !hr.u.ax.property_get.parameter1) {
+		dev_info(&pci_dev->dev,
+			"IRQs not supported by adapter at index %d\n",
+			adapter.adapter->index);
+	} else {
+		irq_supported = 1;
+	}
 
 	/* WARNING can't init mutex in 'adapter'
 	 * and then copy it to adapters[] ?!?!
@@ -398,6 +454,44 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	mutex_init(&adapters[adapter_index].mutex);
 	pci_set_drvdata(pci_dev, &adapters[adapter_index]);
 
+	if (low_latency_mode && irq_supported) {
+		if (!adapter.adapter->irq_query_and_clear) {
+			dev_err(&pci_dev->dev,
+				"no IRQ handler for adapter %d, aborting\n",
+				adapter.adapter->index);
+			goto err;
+		}
+
+		/* Disable IRQ generation on DSP side by setting the rate to 0 */
+		hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+			HPI_ADAPTER_SET_PROPERTY);
+		hm.adapter_index = adapter.adapter->index;
+		hm.u.ax.property_set.property = HPI_ADAPTER_PROPERTY_IRQ_RATE;
+		hm.u.ax.property_set.parameter1 = 0;
+		hm.u.ax.property_set.parameter2 = 0;
+		hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
+		if (hr.error) {
+			HPI_DEBUG_LOG(ERROR,
+				"HPI_ADAPTER_GET_MODE failed, aborting\n");
+			goto err;
+		}
+
+		/* Note: request_irq calls asihpi_isr here */
+		if (request_irq(pci_dev->irq, asihpi_isr, IRQF_SHARED,
+				"asihpi", &adapters[adapter_index])) {
+			dev_err(&pci_dev->dev, "request_irq(%d) failed\n",
+				pci_dev->irq);
+			goto err;
+		}
+
+		adapters[adapter_index].interrupt_mode = 1;
+
+		dev_info(&pci_dev->dev, "using irq %d\n", pci_dev->irq);
+		adapters[adapter_index].irq = pci_dev->irq;
+	} else {
+		dev_info(&pci_dev->dev, "using polled mode\n");
+	}
+
 	dev_info(&pci_dev->dev, "probe succeeded for ASI%04X HPI index %d\n",
 		 adapter.adapter->type, adapter_index);
 
@@ -431,6 +525,15 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev)
 	pa = pci_get_drvdata(pci_dev);
 	pci = pa->adapter->pci;
 
+	/* Disable IRQ generation on DSP side */
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+		HPI_ADAPTER_SET_PROPERTY);
+	hm.adapter_index = pa->adapter->index;
+	hm.u.ax.property_set.property = HPI_ADAPTER_PROPERTY_IRQ_RATE;
+	hm.u.ax.property_set.parameter1 = 0;
+	hm.u.ax.property_set.parameter2 = 0;
+	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
+
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
 		HPI_ADAPTER_DELETE);
 	hm.adapter_index = pa->adapter->index;
@@ -442,6 +545,9 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev)
 			iounmap(pci.ap_mem_base[idx]);
 	}
 
+	if (pa->irq)
+		free_irq(pa->irq, pa);
+
 	if (pa->p_buffer)
 		vfree(pa->p_buffer);
 
diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index d17d017940d8..4e383601b9cf 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -151,6 +151,10 @@ struct hpi_adapter {
 	struct hpi_adapter_obj *adapter;
 	struct snd_card *snd_card;
 
+	int irq;
+	int interrupt_mode;
+	void (*interrupt_callback) (struct hpi_adapter *);
+
 	/* mutex prevents contention for one card
 	   between multiple user programs (via ioctl) */
 	struct mutex mutex;
-- 
cgit v1.2.3


From 5bc91f5b3c732bdb3b9e7cc8bd27969d25015bcd Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:54 +1300
Subject: ALSA: asihpi: Turn off msg/resp logging after DSP has crashed.

Prevents spewing of useless messages if app keeps trying to
access the card.

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpimsgx.c | 15 ++++++++++++---
 1 file changed, 12 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index d4790ddc225c..736f45337fc7 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -35,6 +35,7 @@ static struct pci_device_id asihpi_pci_tbl[] = {
 static struct hpios_spinlock msgx_lock;
 
 static hpi_handler_func *hpi_entry_points[HPI_MAX_ADAPTERS];
+static int logging_enabled = 1;
 
 static hpi_handler_func *hpi_lookup_entry_point_function(const struct hpi_pci
 	*pci_info)
@@ -312,7 +313,9 @@ static void instream_message(struct hpi_message *phm,
 void hpi_send_recv_ex(struct hpi_message *phm, struct hpi_response *phr,
 	void *h_owner)
 {
-	HPI_DEBUG_MESSAGE(DEBUG, phm);
+
+	if (logging_enabled)
+		HPI_DEBUG_MESSAGE(DEBUG, phm);
 
 	if (phm->type != HPI_TYPE_REQUEST) {
 		hpi_init_response(phr, phm->object, phm->function,
@@ -352,8 +355,14 @@ void hpi_send_recv_ex(struct hpi_message *phm, struct hpi_response *phr,
 		hw_entry_point(phm, phr);
 		break;
 	}
-	HPI_DEBUG_RESPONSE(phr);
 
+	if (logging_enabled)
+		HPI_DEBUG_RESPONSE(phr);
+
+	if (phr->error >= HPI_ERROR_DSP_COMMUNICATION) {
+		hpi_debug_level_set(HPI_DEBUG_LEVEL_ERROR);
+		logging_enabled = 0;
+	}
 }
 
 static void adapter_open(struct hpi_message *phm, struct hpi_response *phr)
-- 
cgit v1.2.3


From 12eb0898741870882ca474708e811983d5a5d768 Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:55 +1300
Subject: ALSA: asihpi: Use standard printk helpers

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/asihpi.c | 32 +++++++++++++-------------------
 1 file changed, 13 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index ae29f30547cc..e9273fb2a505 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -380,7 +380,7 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi,
 					  HPI_SOURCENODE_CLOCK_SOURCE, 0, 0, 0,
 					  HPI_CONTROL_SAMPLECLOCK, &h_control);
 		if (err) {
-			snd_printk(KERN_ERR
+			dev_err(&asihpi->pci->dev,
 				"No local sampleclock, err %d\n", err);
 		}
 
@@ -1438,7 +1438,7 @@ static inline int ctl_add(struct snd_card *card, struct snd_kcontrol_new *ctl,
 	if (err < 0)
 		return err;
 	else if (mixer_dump)
-		snd_printk(KERN_INFO "added %s(%d)\n", ctl->name, ctl->index);
+		dev_info(&asihpi->pci->dev, "added %s(%d)\n", ctl->name, ctl->index);
 
 	return 0;
 }
@@ -2652,7 +2652,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 		if (err) {
 			if (err == HPI_ERROR_CONTROL_DISABLED) {
 				if (mixer_dump)
-					snd_printk(KERN_INFO
+					dev_info(&asihpi->pci->dev,
 						   "Disabled HPI Control(%d)\n",
 						   idx);
 				continue;
@@ -2717,9 +2717,8 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 		case HPI_CONTROL_COMPANDER:
 		default:
 			if (mixer_dump)
-				snd_printk(KERN_INFO
-					"Untranslated HPI Control"
-					"(%d) %d %d %d %d %d\n",
+				dev_info(&asihpi->pci->dev,
+					"Untranslated HPI Control (%d) %d %d %d %d %d\n",
 					idx,
 					hpi_ctl.control_type,
 					hpi_ctl.src_node_type,
@@ -2734,7 +2733,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 	if (HPI_ERROR_INVALID_OBJ_INDEX != err)
 		hpi_handle_error(err);
 
-	snd_printk(KERN_INFO "%d mixer controls found\n", idx);
+	dev_info(&asihpi->pci->dev, "%d mixer controls found\n", idx);
 
 	return 0;
 }
@@ -2897,8 +2896,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 				   &card);
 		if (err < 0)
 			return err;
-		snd_printk(KERN_WARNING
-			"**** WARNING **** Adapter index %d->ALSA index %d\n",
+		dev_warn(&pci_dev->dev, "Adapter index %d->ALSA index %d\n",
 			adapter_index, card->number);
 	}
 
@@ -2908,9 +2906,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	asihpi->hpi = hpi;
 	hpi->snd_card = card;
 
-	snd_printk(KERN_INFO "adapter ID=%4X index=%d\n",
-			asihpi->hpi->adapter->type, adapter_index);
-
 	err = hpi_adapter_get_property(adapter_index,
 		HPI_ADAPTER_PROPERTY_CAPS1,
 		NULL, &asihpi->support_grouping);
@@ -2968,22 +2963,21 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 		asihpi->in_min_chans = 1;
 	}
 
-	snd_printk(KERN_INFO "update_interval_frames: %d",
-		asihpi->update_interval_frames);
-	snd_printk(KERN_INFO "Has dma:%d, grouping:%d, mrx:%d\n",
+	dev_info(&pci_dev->dev, "Has dma:%d, grouping:%d, mrx:%d, uif:%d\n",
 			asihpi->can_dma,
 			asihpi->support_grouping,
-			asihpi->support_mrx
+			asihpi->support_mrx,
+			asihpi->update_interval_frames
 	      );
 
 	err = snd_card_asihpi_pcm_new(asihpi, 0);
 	if (err < 0) {
-		snd_printk(KERN_ERR "pcm_new failed\n");
+		dev_err(&pci_dev->dev, "pcm_new failed\n");
 		goto __nodev;
 	}
 	err = snd_card_asihpi_mixer_new(asihpi);
 	if (err < 0) {
-		snd_printk(KERN_ERR "mixer_new failed\n");
+		dev_err(&pci_dev->dev, "mixer_new failed\n");
 		goto __nodev;
 	}
 
@@ -3015,7 +3009,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	}
 __nodev:
 	snd_card_free(card);
-	snd_printk(KERN_ERR "snd_asihpi_probe error %d\n", err);
+	dev_err(&pci_dev->dev, "snd_asihpi_probe error %d\n", err);
 	return err;
 
 }
-- 
cgit v1.2.3


From dc612838eac746b11bb4e5d923dafeea0ba7e81b Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:56 +1300
Subject: ALSA: asihpi: don't fail probe if adapter mode read fails

Only determining if low latency mode is enabled. Failure
indicates adapter has no modes

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpioctl.c | 13 ++++++-------
 1 file changed, 6 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 9454932fc9c0..e457eb80658b 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -424,14 +424,13 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	hm.adapter_index = adapter.adapter->index;
 	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
 
-	if (hr.error) {
-		HPI_DEBUG_LOG(ERROR,
-			"HPI_ADAPTER_GET_MODE failed, aborting\n");
-		goto err;
-	}
-
-	if (hr.u.ax.mode.adapter_mode == HPI_ADAPTER_MODE_LOW_LATENCY)
+	if (!hr.error
+		&& hr.u.ax.mode.adapter_mode == HPI_ADAPTER_MODE_LOW_LATENCY)
 		low_latency_mode = 1;
+	else
+		dev_info(&pci_dev->dev,
+			"Adapter at index %d is not in low latency mode\n",
+			adapter.adapter->index);
 
 	/* Check if IRQs are supported */
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
-- 
cgit v1.2.3


From 51e6f47dd2e3463dac6f37128fd7b7cb40c500de Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Date: Thu, 20 Nov 2014 16:22:57 +1300
Subject: ALSA: asihpi: used parts of message/response are zeroed before use

Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpimsginit.c | 30 ++++++++++++++++++++----------
 1 file changed, 20 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpimsginit.c b/sound/pci/asihpi/hpimsginit.c
index 032d563e3708..7eb617175fde 100644
--- a/sound/pci/asihpi/hpimsginit.c
+++ b/sound/pci/asihpi/hpimsginit.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -37,11 +37,15 @@ static u16 gwSSX2_bypass;
 static void hpi_init_message(struct hpi_message *phm, u16 object,
 	u16 function)
 {
-	memset(phm, 0, sizeof(*phm));
+	u16 size;
+
 	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX))
-		phm->size = msg_size[object];
+		size = msg_size[object];
 	else
-		phm->size = sizeof(*phm);
+		size = sizeof(*phm);
+
+	memset(phm, 0, size);
+	phm->size = size;
 
 	if (gwSSX2_bypass)
 		phm->type = HPI_TYPE_SSX2BYPASS_MESSAGE;
@@ -60,12 +64,16 @@ static void hpi_init_message(struct hpi_message *phm, u16 object,
 void hpi_init_response(struct hpi_response *phr, u16 object, u16 function,
 	u16 error)
 {
-	memset(phr, 0, sizeof(*phr));
-	phr->type = HPI_TYPE_RESPONSE;
+	u16 size;
+
 	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX))
-		phr->size = res_size[object];
+		size = res_size[object];
 	else
-		phr->size = sizeof(*phr);
+		size = sizeof(*phr);
+
+	memset(phr, 0, sizeof(*phr));
+	phr->size = size;
+	phr->type = HPI_TYPE_RESPONSE;
 	phr->object = object;
 	phr->function = function;
 	phr->error = error;
@@ -86,7 +94,7 @@ void hpi_init_message_response(struct hpi_message *phm,
 static void hpi_init_messageV1(struct hpi_message_header *phm, u16 size,
 	u16 object, u16 function)
 {
-	memset(phm, 0, sizeof(*phm));
+	memset(phm, 0, size);
 	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) {
 		phm->size = size;
 		phm->type = HPI_TYPE_REQUEST;
@@ -100,7 +108,9 @@ static void hpi_init_messageV1(struct hpi_message_header *phm, u16 size,
 void hpi_init_responseV1(struct hpi_response_header *phr, u16 size,
 	u16 object, u16 function)
 {
-	memset(phr, 0, sizeof(*phr));
+	(void)object;
+	(void)function;
+	memset(phr, 0, size);
 	phr->size = size;
 	phr->version = 1;
 	phr->type = HPI_TYPE_RESPONSE;
-- 
cgit v1.2.3


From 69eba10e606a80665f8573221fec589430d9d1cb Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Thu, 27 Nov 2014 01:34:43 +0300
Subject: ALSA: hda - using uninitialized data

In olden times the snd_hda_param_read() function always set "*start_id"
but in 2007 we introduced a new return and it causes uninitialized data
bugs in a couple of the callers: print_codec_info() and
hdmi_parse_codec().

Fixes: e8a7f136f5ed ('[ALSA] hda-intel - Improve HD-audio codec probing robustness')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b2d58998dbdd..2fe86d2e1b09 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -346,8 +346,10 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
 	unsigned int parm;
 
 	parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT);
-	if (parm == -1)
+	if (parm == -1) {
+		*start_id = 0;
 		return 0;
+	}
 	*start_id = (parm >> 16) & 0x7fff;
 	return (int)(parm & 0x7fff);
 }
-- 
cgit v1.2.3


From 37e661ee10c6d0d1310c62b3d29ae9a63073ac5d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Nov 2014 11:28:07 +0100
Subject: ALSA: hda - Add AZX_DCAPS_SNOOP_OFF (and refactor snoop setup)

Add a new driver_caps bit, AZX_DCAPS_SNOOP_OFF, to set the snoop off
as default.  This new bit is used for the checks in
azx_check_snoop_available().  Most of case-switches are replaced with
the new dcaps in each entry.

While working on it, for avoiding to spend more bits, combine three
bits AZX_DCAPS_SNOOP_SCH, AZX_DCAPS_SNOOP_ATI and
AZX_DCAPS_SNOOP_NVIDIA bits into a flat type of two bits.  This
reduces the bits usages, and assign AZX_DCAPS_OFF to this empty bit
now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 71 +++++++++++++++++++++++++----------------------
 sound/pci/hda/hda_priv.h  | 12 ++++++--
 2 files changed, 47 insertions(+), 36 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 966e6f98892d..633020de9bd2 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -272,42 +272,51 @@ enum {
 	AZX_NUM_DRIVERS, /* keep this as last entry */
 };
 
+#define azx_get_snoop_type(chip) \
+	(((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10)
+#define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10)
+
 /* quirks for Intel PCH */
 #define AZX_DCAPS_INTEL_PCH_NOPM \
-	(AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \
-	 AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_REVERSE_ASSIGN)
+	(AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
+	 AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH))
 
 #define AZX_DCAPS_INTEL_PCH \
 	(AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME)
 
 #define AZX_DCAPS_INTEL_HASWELL \
-	(AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \
-	 AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME | \
-	 AZX_DCAPS_I915_POWERWELL)
+	(AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
+	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
+	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
 /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */
 #define AZX_DCAPS_INTEL_BROADWELL \
-	(AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \
-	 AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_PM_RUNTIME | \
-	 AZX_DCAPS_I915_POWERWELL)
+	(AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_POSFIX_LPIB |\
+	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
+	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
-	(AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \
-	 AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB)
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
+	 AZX_DCAPS_SNOOP_TYPE(ATI))
 
 /* quirks for ATI/AMD HDMI */
 #define AZX_DCAPS_PRESET_ATI_HDMI \
 	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB)
 
+/* quirks for ATI HDMI with snoop off */
+#define AZX_DCAPS_PRESET_ATI_HDMI_NS \
+	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
+
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
-	(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
-	 AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT |\
-	 AZX_DCAPS_CORBRP_SELF_CLEAR)
+	(AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | AZX_DCAPS_ALIGN_BUFSIZE |\
+	 AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\
+	 AZX_DCAPS_SNOOP_TYPE(NVIDIA))
 
 #define AZX_DCAPS_PRESET_CTHDA \
-	(AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY)
+	(AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB |\
+	 AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF)
 
 /*
  * VGA-switcher support
@@ -436,6 +445,8 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg,
 
 static void azx_init_pci(struct azx *chip)
 {
+	int snoop_type = azx_get_snoop_type(chip);
+
 	/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
 	 * TCSEL == Traffic Class Select Register, which sets PCI express QOS
 	 * Ensuring these bits are 0 clears playback static on some HD Audio
@@ -450,7 +461,7 @@ static void azx_init_pci(struct azx *chip)
 	/* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio,
 	 * we need to enable snoop.
 	 */
-	if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) {
+	if (snoop_type == AZX_SNOOP_TYPE_ATI) {
 		dev_dbg(chip->card->dev, "Setting ATI snoop: %d\n",
 			azx_snoop(chip));
 		update_pci_byte(chip->pci,
@@ -459,7 +470,7 @@ static void azx_init_pci(struct azx *chip)
 	}
 
 	/* For NVIDIA HDA, enable snoop */
-	if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) {
+	if (snoop_type == AZX_SNOOP_TYPE_NVIDIA) {
 		dev_dbg(chip->card->dev, "Setting Nvidia snoop: %d\n",
 			azx_snoop(chip));
 		update_pci_byte(chip->pci,
@@ -474,7 +485,7 @@ static void azx_init_pci(struct azx *chip)
 	}
 
 	/* Enable SCH/PCH snoop if needed */
-	if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) {
+	if (snoop_type == AZX_SNOOP_TYPE_SCH) {
 		unsigned short snoop;
 		pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
 		if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) ||
@@ -1361,8 +1372,8 @@ static void azx_check_snoop_available(struct azx *chip)
 {
 	bool snoop = chip->snoop;
 
-	switch (chip->driver_type) {
-	case AZX_DRIVER_VIA:
+	if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE &&
+	    chip->driver_type == AZX_DRIVER_VIA) {
 		/* force to non-snoop mode for a new VIA controller
 		 * when BIOS is set
 		 */
@@ -1372,17 +1383,11 @@ static void azx_check_snoop_available(struct azx *chip)
 			if (!(val & 0x80) && chip->pci->revision == 0x30)
 				snoop = false;
 		}
-		break;
-	case AZX_DRIVER_ATIHDMI_NS:
-		/* new ATI HDMI requires non-snoop */
-		snoop = false;
-		break;
-	case AZX_DRIVER_CTHDA:
-	case AZX_DRIVER_CMEDIA:
-		snoop = false;
-		break;
 	}
 
+	if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF)
+		snoop = false;
+
 	if (snoop != chip->snoop) {
 		dev_info(chip->card->dev, "Force to %s mode\n",
 			 snoop ? "snoop" : "non-snoop");
@@ -2116,13 +2121,13 @@ static const struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x1002, 0xaa98),
 	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
 	{ PCI_DEVICE(0x1002, 0x9902),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
 	{ PCI_DEVICE(0x1002, 0xaaa0),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
 	{ PCI_DEVICE(0x1002, 0xaaa8),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
 	{ PCI_DEVICE(0x1002, 0xaab0),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
 	/* VIA VT8251/VT8237A */
 	{ PCI_DEVICE(0x1106, 0x3288),
 	  .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA },
@@ -2169,7 +2174,7 @@ static const struct pci_device_id azx_ids[] = {
 	/* CM8888 */
 	{ PCI_DEVICE(0x13f6, 0x5011),
 	  .driver_data = AZX_DRIVER_CMEDIA |
-	  AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB },
+	  AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_SNOOP_OFF },
 	/* Vortex86MX */
 	{ PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC },
 	/* VMware HDAudio */
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
index 949cd437eeb2..602536c2147d 100644
--- a/sound/pci/hda/hda_priv.h
+++ b/sound/pci/hda/hda_priv.h
@@ -152,9 +152,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 /* bits 0-7 are used for indicating driver type */
 #define AZX_DCAPS_NO_TCSEL	(1 << 8)	/* No Intel TCSEL bit */
 #define AZX_DCAPS_NO_MSI	(1 << 9)	/* No MSI support */
-#define AZX_DCAPS_ATI_SNOOP	(1 << 10)	/* ATI snoop enable */
-#define AZX_DCAPS_NVIDIA_SNOOP	(1 << 11)	/* Nvidia snoop enable */
-#define AZX_DCAPS_SCH_SNOOP	(1 << 12)	/* SCH/PCH snoop enable */
+#define AZX_DCAPS_SNOOP_MASK	(3 << 10)	/* snoop type mask */
+#define AZX_DCAPS_SNOOP_OFF	(1 << 12)	/* snoop default off */
 #define AZX_DCAPS_RIRB_DELAY	(1 << 13)	/* Long delay in read loop */
 #define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14)	/* Put a delay before read */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
@@ -172,6 +171,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define AZX_DCAPS_I915_POWERWELL (1 << 27)	/* HSW i915 powerwell support */
 #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
 
+enum {
+	AZX_SNOOP_TYPE_NONE ,
+	AZX_SNOOP_TYPE_SCH,
+	AZX_SNOOP_TYPE_ATI,
+	AZX_SNOOP_TYPE_NVIDIA,
+};
+
 /* HD Audio class code */
 #define PCI_CLASS_MULTIMEDIA_HD_AUDIO	0x0403
 
-- 
cgit v1.2.3


From 7c7320157a37ed459b59e2f6b53b73780b12ad80 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Nov 2014 12:54:16 +0100
Subject: ALSA: hda - Allow forcibly enabling/disabling snoop

User can pass snoop option to enable/disable the snoop behavior, but
currently azx_check_snoop_available() always turns it off for some
devices.  For better debuggability, change the parameter as bint, and
allow user to enable/disable forcibly the snoop when specified via the
module option.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 33 ++++++++++++++++++---------------
 1 file changed, 18 insertions(+), 15 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 633020de9bd2..728663d6746f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -196,8 +196,8 @@ MODULE_PARM_DESC(align_buffer_size,
 		"Force buffer and period sizes to be multiple of 128 bytes.");
 
 #ifdef CONFIG_X86
-static bool hda_snoop = true;
-module_param_named(snoop, hda_snoop, bool, 0444);
+static int hda_snoop = -1;
+module_param_named(snoop, hda_snoop, bint, 0444);
 MODULE_PARM_DESC(snoop, "Enable/disable snooping");
 #else
 #define hda_snoop		true
@@ -1370,29 +1370,33 @@ static void check_msi(struct azx *chip)
 /* check the snoop mode availability */
 static void azx_check_snoop_available(struct azx *chip)
 {
-	bool snoop = chip->snoop;
+	int snoop = hda_snoop;
 
+	if (snoop >= 0) {
+		dev_info(chip->card->dev, "Force to %s mode by module option\n",
+			 snoop ? "snoop" : "non-snoop");
+		chip->snoop = snoop;
+		return;
+	}
+
+	snoop = true;
 	if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE &&
 	    chip->driver_type == AZX_DRIVER_VIA) {
 		/* force to non-snoop mode for a new VIA controller
 		 * when BIOS is set
 		 */
-		if (snoop) {
-			u8 val;
-			pci_read_config_byte(chip->pci, 0x42, &val);
-			if (!(val & 0x80) && chip->pci->revision == 0x30)
-				snoop = false;
-		}
+		u8 val;
+		pci_read_config_byte(chip->pci, 0x42, &val);
+		if (!(val & 0x80) && chip->pci->revision == 0x30)
+			snoop = false;
 	}
 
 	if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF)
 		snoop = false;
 
-	if (snoop != chip->snoop) {
-		dev_info(chip->card->dev, "Force to %s mode\n",
-			 snoop ? "snoop" : "non-snoop");
-		chip->snoop = snoop;
-	}
+	chip->snoop = snoop;
+	if (!snoop)
+		dev_info(chip->card->dev, "Force to non-snoop mode\n");
 }
 
 static void azx_probe_work(struct work_struct *work)
@@ -1452,7 +1456,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 	check_probe_mask(chip, dev);
 
 	chip->single_cmd = single_cmd;
-	chip->snoop = hda_snoop;
 	azx_check_snoop_available(chip);
 
 	if (bdl_pos_adj[dev] < 0) {
-- 
cgit v1.2.3


From 87164cc5723329565089a999b6671bd214caf0a0 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Tue, 2 Dec 2014 18:05:32 +0100
Subject: ALSA: asihpi: Deletion of an unnecessary check before the function
 call "vfree"

The vfree() function performs also input parameter validation. Thus the test
around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpioctl.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index e457eb80658b..6aa677e60555 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -547,8 +547,7 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev)
 	if (pa->irq)
 		free_irq(pa->irq, pa);
 
-	if (pa->p_buffer)
-		vfree(pa->p_buffer);
+	vfree(pa->p_buffer);
 
 	if (1)
 		dev_info(&pci_dev->dev,
-- 
cgit v1.2.3


From 057a4a55e703038d22bc9f2bcf8b02dc35850e16 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Tue, 2 Dec 2014 18:34:45 +0100
Subject: ALSA: echoaudio: Deletion of a check before
 release_and_free_resource()

The release_and_free_resource() function tests whether its argument is NULL
and then returns immediately. Thus the test around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 60e40034b991..21228adaa70c 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1875,8 +1875,7 @@ static int snd_echo_free(struct echoaudio *chip)
 	if (chip->dsp_registers)
 		iounmap(chip->dsp_registers);
 
-	if (chip->iores)
-		release_and_free_resource(chip->iores);
+	release_and_free_resource(chip->iores);
 
 
 	pci_disable_device(chip->pci);
-- 
cgit v1.2.3


From 5c34fdf48b9522ca87372b1fae19de8f93ffd130 Mon Sep 17 00:00:00 2001
From: Markus Elfring <elfring@users.sourceforge.net>
Date: Tue, 2 Dec 2014 18:52:21 +0100
Subject: ALSA: trident: Deletion of a check before snd_util_memhdr_free()

The snd_util_memhdr_free() function tests whether its argument is NULL
and then returns immediately. Thus the test around the call is not needed.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/trident/trident_main.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index da875dced2ef..57cd757acfe7 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3702,8 +3702,7 @@ static int snd_trident_free(struct snd_trident *trident)
 		free_irq(trident->irq, trident);
 	if (trident->tlb.buffer.area) {
 		outl(0, TRID_REG(trident, NX_TLBC));
-		if (trident->tlb.memhdr)
-			snd_util_memhdr_free(trident->tlb.memhdr);
+		snd_util_memhdr_free(trident->tlb.memhdr);
 		if (trident->tlb.silent_page.area)
 			snd_dma_free_pages(&trident->tlb.silent_page);
 		vfree(trident->tlb.shadow_entries);
-- 
cgit v1.2.3


From b42b4afb7482f1c079c82af824a7fe750590f438 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 3 Dec 2014 09:47:20 +0100
Subject: ALSA: hda - Define the DCAPS preset for the old Intel chipsets

Just for improving readability.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 37 ++++++++++++++++++++-----------------
 1 file changed, 20 insertions(+), 17 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index fc7aff0eb562..53e43d111a3b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -276,6 +276,10 @@ enum {
 	(((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10)
 #define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10)
 
+/* quirks for old Intel chipsets */
+#define AZX_DCAPS_INTEL_ICH \
+	(AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE)
+
 /* quirks for Intel PCH */
 #define AZX_DCAPS_INTEL_PCH_NOPM \
 	(AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
@@ -2054,31 +2058,30 @@ static const struct pci_device_id azx_ids[] = {
 	/* Braswell */
 	{ PCI_DEVICE(0x8086, 0x2284),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
-	/* ICH */
+	/* ICH6 */
 	{ PCI_DEVICE(0x8086, 0x2668),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH6 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ICH7 */
 	{ PCI_DEVICE(0x8086, 0x27d8),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH7 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ESB2 */
 	{ PCI_DEVICE(0x8086, 0x269a),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ESB2 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ICH8 */
 	{ PCI_DEVICE(0x8086, 0x284b),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH8 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ICH9 */
 	{ PCI_DEVICE(0x8086, 0x293e),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH9 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ICH9 */
 	{ PCI_DEVICE(0x8086, 0x293f),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH9 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ICH10 */
 	{ PCI_DEVICE(0x8086, 0x3a3e),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH10 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
+	/* ICH10 */
 	{ PCI_DEVICE(0x8086, 0x3a6e),
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
-	  AZX_DCAPS_BUFSIZE },  /* ICH10 */
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH },
 	/* Generic Intel */
 	{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
-- 
cgit v1.2.3


From 103884a351a221553095c509a1dbbbf7d4fd9b05 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 3 Dec 2014 09:56:20 +0100
Subject: ALSA: hda - Drop AZX_DCAPS_ALIGN_BUFSIZE

We introduced AZX_DCAPS_ALIGN_BUFSIZE to explicity show that the
controller needs the alignment, with a slight hope that the buffer
size alignment will be disabled as default in future.  But the reality
tells that most chips need the buffer size alignment, and it'll be
likely enabled in future, too.

This patch drops AZX_DCAPS_ALIGN_BUFSIZE to give back one more
precious DCAPS bit for future use.  At the same time, rename
AZX_DCAPS_BUFSIZE with AZX_DCAPS_NO_ALIGN_BUFSIZE for avoiding
confusion.

AZX_DCAPS_ALIGN_BUFSIZE are still kept (but commented out) in each
DCAPS presets for a purpose as markers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 16 +++++++---------
 sound/pci/hda/hda_priv.h  |  4 ++--
 2 files changed, 9 insertions(+), 11 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 53e43d111a3b..5ac0d39d59bc 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -278,24 +278,24 @@ enum {
 
 /* quirks for old Intel chipsets */
 #define AZX_DCAPS_INTEL_ICH \
-	(AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE)
+	(AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE)
 
 /* quirks for Intel PCH */
 #define AZX_DCAPS_INTEL_PCH_NOPM \
-	(AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
+	(AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
 	 AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH))
 
 #define AZX_DCAPS_INTEL_PCH \
 	(AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME)
 
 #define AZX_DCAPS_INTEL_HASWELL \
-	(AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
+	(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\
 	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
 	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
 /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */
 #define AZX_DCAPS_INTEL_BROADWELL \
-	(AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_POSFIX_LPIB |\
+	(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\
 	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
 	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
@@ -315,7 +315,7 @@ enum {
 
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
-	(AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | AZX_DCAPS_ALIGN_BUFSIZE |\
+	(AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | /*AZX_DCAPS_ALIGN_BUFSIZE |*/ \
 	 AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\
 	 AZX_DCAPS_SNOOP_TYPE(NVIDIA))
 
@@ -1568,10 +1568,8 @@ static int azx_first_init(struct azx *chip)
 	if (align_buffer_size >= 0)
 		chip->align_buffer_size = !!align_buffer_size;
 	else {
-		if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
+		if (chip->driver_caps & AZX_DCAPS_NO_ALIGN_BUFSIZE)
 			chip->align_buffer_size = 0;
-		else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE)
-			chip->align_buffer_size = 1;
 		else
 			chip->align_buffer_size = 1;
 	}
@@ -2086,7 +2084,7 @@ static const struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
-	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE },
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_NO_ALIGN_BUFSIZE },
 	/* ATI SB 450/600/700/800/900 */
 	{ PCI_DEVICE(0x1002, 0x437b),
 	  .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
index a09703a2b2c1..aa484fdf4338 100644
--- a/sound/pci/hda/hda_priv.h
+++ b/sound/pci/hda/hda_priv.h
@@ -162,8 +162,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
-#define AZX_DCAPS_BUFSIZE	(1 << 21)	/* no buffer size alignment */
-#define AZX_DCAPS_ALIGN_BUFSIZE	(1 << 22)	/* buffer size alignment */
+#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21)	/* no buffer size alignment */
+/* 22 unused */
 #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23)	/* BDLE in 4k boundary */
 #define AZX_DCAPS_REVERSE_ASSIGN (1 << 24)	/* Assign devices in reverse order */
 #define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
-- 
cgit v1.2.3


From 5395103dcc709d87f08edaecb786fc37781f3b22 Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Wed, 3 Dec 2014 09:59:31 -0800
Subject: ALSA: ctxfi: Neaten get_daio_rsc

Move the pointer declarations into the blocks that use them.
Neaten the kfree calls when the _init functions fail.

Trivially reduces object size (defconfig x86-64)

$ size sound/pci/ctxfi/ctdaio.o.*
   text	   data	    bss	    dec	    hex	filename
   5287	    224	      0	   5511	   1587	sound/pci/ctxfi/ctdaio.o.new
   5319	    224	      0	   5543	   15a7	sound/pci/ctxfi/ctdaio.o.old

Signed-off-by: Joe Perches <joe@perches.com>
Noticed-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctdaio.c | 30 +++++++++++++-----------------
 1 file changed, 13 insertions(+), 17 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index c1c3f8816fff..9b87dd28de83 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -528,8 +528,6 @@ static int get_daio_rsc(struct daio_mgr *mgr,
 			struct daio **rdaio)
 {
 	int err;
-	struct dai *dai = NULL;
-	struct dao *dao = NULL;
 	unsigned long flags;
 
 	*rdaio = NULL;
@@ -544,27 +542,30 @@ static int get_daio_rsc(struct daio_mgr *mgr,
 		return err;
 	}
 
+	err = -ENOMEM;
 	/* Allocate mem for daio resource */
 	if (desc->type <= DAIO_OUT_MAX) {
-		dao = kzalloc(sizeof(*dao), GFP_KERNEL);
-		if (!dao) {
-			err = -ENOMEM;
+		struct dao *dao = kzalloc(sizeof(*dao), GFP_KERNEL);
+		if (!dao)
 			goto error;
-		}
+
 		err = dao_rsc_init(dao, desc, mgr);
-		if (err)
+		if (err) {
+			kfree(dao);
 			goto error;
+		}
 
 		*rdaio = &dao->daio;
 	} else {
-		dai = kzalloc(sizeof(*dai), GFP_KERNEL);
-		if (!dai) {
-			err = -ENOMEM;
+		struct dai *dai = kzalloc(sizeof(*dai), GFP_KERNEL);
+		if (!dai)
 			goto error;
-		}
+
 		err = dai_rsc_init(dai, desc, mgr);
-		if (err)
+		if (err) {
+			kfree(dai);
 			goto error;
+		}
 
 		*rdaio = &dai->daio;
 	}
@@ -575,11 +576,6 @@ static int get_daio_rsc(struct daio_mgr *mgr,
 	return 0;
 
 error:
-	if (dao)
-		kfree(dao);
-	else if (dai)
-		kfree(dai);
-
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	daio_mgr_put_rsc(&mgr->mgr, desc->type);
 	spin_unlock_irqrestore(&mgr->mgr_lock, flags);
-- 
cgit v1.2.3


From fedb2245cbb8d823e449ebdd48ba9bb35c071ce0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Nov 2014 07:11:38 +0100
Subject: ALSA: hda - Fix built-in mic at resume on Lenovo Ideapad S210

The built-in mic boost volume gets almost muted after suspend/resume
on Lenovo Ideapad S210.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=88121
Reported-and-tested-by: Roman Kagan <rkagan@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b118a5be18df..c5ad83e4e0c7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4887,6 +4887,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
 	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC),
-- 
cgit v1.2.3


From f62f5eff3d40a56ad1cf0d81a6cac8dd8743e8a1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 9 Dec 2014 19:58:53 +0100
Subject: ALSA: hda - Add EAPD fixup for ASUS Z99He laptop

The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A
codec like another ASUS machine.

Reported-and-tested-by: Dmitry V. Zimin <pfzim@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 06275f8807a8..4714ff92f15e 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = {
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD),
 	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
 	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
-- 
cgit v1.2.3


From 9faa73f06ec2408cf86d20758d879b8d928ab30a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 10 Dec 2014 13:58:37 +0100
Subject: ALSA: hda - Add "eapd" model string for AD1986A codec

Also update the documentation to the latest state.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/HD-Audio-Models.txt | 8 ++------
 sound/pci/hda/patch_analog.c                 | 1 +
 2 files changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index a5e754714344..5a3163cac6c3 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -113,14 +113,10 @@ AD1984
 
 AD1986A
 =======
-  6stack	6-jack, separate surrounds (default)
   3stack	3-stack, shared surrounds
   laptop	2-channel only (FSC V2060, Samsung M50)
-  laptop-eapd	2-channel with EAPD (ASUS A6J)
-  laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
-  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
-  samsung	2-channel with EAPD (Samsung R65)
-  samsung-p50	2-channel with HP-automute (Samsung P50)
+  laptop-imic	2-channel with built-in mic
+  eapd		Turn on EAPD constantly
 
 AD1988/AD1988B/AD1989A/AD1989B
 ==============================
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 4714ff92f15e..c81b715d6c98 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -352,6 +352,7 @@ static const struct hda_model_fixup ad1986a_fixup_models[] = {
 	{ .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" },
 	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" },
 	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */
+	{ .id = AD1986A_FIXUP_EAPD, .name = "eapd" },
 	{}
 };
 
-- 
cgit v1.2.3


From feabb67e0eb0f509b530a56756ea44364843faea Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Wed, 10 Dec 2014 16:06:46 +0300
Subject: ALSA: lola: NULL dereference on probe failure

"card" is NULL if snd_card_new() fails.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/lola/lola.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index a75c8dc66dec..4cf4be5ef82a 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -719,7 +719,7 @@ static int lola_probe(struct pci_dev *pci,
 	err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
 			   0, &card);
 	if (err < 0) {
-		dev_err(card->dev, "Error creating card!\n");
+		dev_err(&pci->dev, "Error creating card!\n");
 		return err;
 	}
 
-- 
cgit v1.2.3


From 6e1d7a51392f06899bd7b693f28ac60fa1e00032 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Wed, 10 Dec 2014 16:26:21 +0300
Subject: ALSA: pcxhr: NULL dereference on probe failure

"card" is NULL if snd_card_new() fails.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index a60293015267..c6092e48ceb6 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1638,7 +1638,7 @@ static int pcxhr_probe(struct pci_dev *pci,
 				   0, &card);
 
 		if (err < 0) {
-			dev_err(card->dev, "cannot allocate the card %d\n", i);
+			dev_err(&pci->dev, "cannot allocate the card %d\n", i);
 			pcxhr_free(mgr);
 			return err;
 		}
-- 
cgit v1.2.3