Merge tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've got many code additions at this cycle as a result of quite a few
  new drivers. Below are highlights:

  Core stuff:
   - Fix the long-standing issue with the device registration order; the
     control device is now registered at last
   - PCM locking code cleanups for RT kernels
   - Fixes for possible races in ALSA timer resolution accesses
   - TLV offset definitions in uapi

  ASoC:
   - Many fixes for the topology stuff, including fixes for v4 ABI
     compatibility
   - Lots of cleanups / quirks for Intel platforms based on Realtek
     CODECs
   - Continued componentization works, removing legacy CODEC stuff
   - Conversion of OMAP DMA to the new, more standard SDMA-PCM driver
   - Fixes and updates to Cirrus Logic SoC drivers
   - New Qualcomm DSP support
   - New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
     MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306
     and RT5668 and TI TSCS454

  HD-audio:
   - Finally better support for some CA0132 boards, allowing Windows
     firmware
   - HP Spectre x360 support along with a bulk of COEF stuff
   - Blacklisting power save default some known boards reported on
     Fedora

  USB-audio:
   - Continued improvements on UAC3 support; now BADD is supported
   - Fixes / improvements for Dell WD15 dock
   - Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co

  Others:
   - New Xen sound frontend driver support
   - Cache implementation and other improvements for FireWire DICE
   - Conversions to octal permissions in allover places"

* tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (386 commits)
  ASoC: dapm: delete dapm_kcontrol_data paths list before freeing it
  ALSA: usb-audio: remove redundant check on err
  ASoC: topology: Move skl-tplg-interface.h to uapi
  ASoC: topology: Move v4 manifest header data structures to uapi
  ASoC: topology: Improve backwards compatibility with v4 topology files
  ALSA: pci/hda: Remove unused, broken, header file
  ASoC: TSCS454: Add Support
  ASoC: Intel: kbl: Move codec sysclk config to codec_init function
  ASoC: simple-card: set cpu dai clk in hw_params
  ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
  ALSA: oxygen: use match_string() helper
  ASoC: dapm: use match_string() helper
  ASoC: max98095: use match_string() helper
  ASoC: max98088: use match_string() helper
  ASoC: Intel: bytcr_rt5651: Set card long_name based on quirks
  ASoC: mt6797-mt6351: add hostless phone call path
  ASoC: mt6797: add Hostless DAI
  ASoC: mt6797: add PCM interface
  ASoC: mediatek: export mtk-afe symbols as needed
  ASoC: codecs: PCM1789: include gpio/consumer.h
  ...

9 files changed, 66 insertions, 95 deletions

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index da8fd98c7f51..8f43110373b8 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -1,12 +1,8 @@
-/*
- * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
 
 #include <linux/clk.h>
 #include <linux/dmaengine.h>
@@ -226,6 +222,12 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned long clk_rate;
 	int ret;
 
+	if (freq == 0) {
+		dev_err(dai->dev, "%sput freq of HCK%c should not be 0Hz\n",
+			in ? "in" : "out", tx ? 'T' : 'R');
+		return -EINVAL;
+	}
+
 	/* Bypass divider settings if the requirement doesn't change */
 	if (freq == esai_priv->hck_rate[tx] && dir == esai_priv->hck_dir[tx])
 		return 0;
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 5e793bbb6b02..f873588d9045 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -1,13 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC
  *
  * Copyright (C) 2014 Freescale Semiconductor, Inc.
  *
  * Author: Nicolin Chen <Guangyu.Chen@freescale.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
  */
 
 #ifndef _FSL_ESAI_DAI_H
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 18e5ce81527d..4163f2cfc06f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1,14 +1,8 @@
-/*
- * Freescale ALSA SoC Digital Audio Interface (SAI) driver.
- *
- * Copyright 2012-2015 Freescale Semiconductor, Inc.
- *
- * This program is free software, you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation, either version 2 of the License, or(at your
- * option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Freescale ALSA SoC Digital Audio Interface (SAI) driver.
+//
+// Copyright 2012-2015 Freescale Semiconductor, Inc.
 
 #include <linux/clk.h>
 #include <linux/delay.h>
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index d9ed7be8cb34..24cb156bf995 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -1,9 +1,6 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * Copyright 2012-2013 Freescale Semiconductor, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __FSL_SAI_H
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 4f7469c1864c..9b59d87b61bf 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1,17 +1,13 @@
-/*
- * Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Copyright (C) 2013 Freescale Semiconductor, Inc.
- *
- * Based on stmp3xxx_spdif_dai.c
- * Vladimir Barinov <vbarinov@embeddedalley.com>
- * Copyright 2008 SigmaTel, Inc
- * Copyright 2008 Embedded Alley Solutions, Inc
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program  is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Copyright (C) 2013 Freescale Semiconductor, Inc.
+//
+// Based on stmp3xxx_spdif_dai.c
+// Vladimir Barinov <vbarinov@embeddedalley.com>
+// Copyright 2008 SigmaTel, Inc
+// Copyright 2008 Embedded Alley Solutions, Inc
 
 #include <linux/bitrev.h>
 #include <linux/clk.h>
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
index 00bd3514c610..7666dabaccfd 100644
--- a/sound/soc/fsl/fsl_spdif.h
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -1,3 +1,4 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * fsl_spdif.h - ALSA S/PDIF interface for the Freescale i.MX SoC
  *
@@ -8,10 +9,6 @@
  * Based on fsl_ssi.h
  * Author: Timur Tabi <timur@freescale.com>
  * Copyright 2007-2008 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program  is licensed "as is" without any warranty of any
- * kind, whether express or implied.
  */
 
 #ifndef _FSL_SPDIF_DAI_H
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 89df2d9f63d7..0a648229e643 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1,34 +1,29 @@
-/*
- * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2007-2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- *
- *
- * Some notes why imx-pcm-fiq is used instead of DMA on some boards:
- *
- * The i.MX SSI core has some nasty limitations in AC97 mode. While most
- * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
- * one FIFO which combines all valid receive slots. We cannot even select
- * which slots we want to receive. The WM9712 with which this driver
- * was developed with always sends GPIO status data in slot 12 which
- * we receive in our (PCM-) data stream. The only chance we have is to
- * manually skip this data in the FIQ handler. With sampling rates different
- * from 48000Hz not every frame has valid receive data, so the ratio
- * between pcm data and GPIO status data changes. Our FIQ handler is not
- * able to handle this, hence this driver only works with 48000Hz sampling
- * rate.
- * Reading and writing AC97 registers is another challenge. The core
- * provides us status bits when the read register is updated with *another*
- * value. When we read the same register two times (and the register still
- * contains the same value) these status bits are not set. We work
- * around this by not polling these bits but only wait a fixed delay.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2007-2010 Freescale Semiconductor, Inc.
+//
+// Some notes why imx-pcm-fiq is used instead of DMA on some boards:
+//
+// The i.MX SSI core has some nasty limitations in AC97 mode. While most
+// sane processor vendors have a FIFO per AC97 slot, the i.MX has only
+// one FIFO which combines all valid receive slots. We cannot even select
+// which slots we want to receive. The WM9712 with which this driver
+// was developed with always sends GPIO status data in slot 12 which
+// we receive in our (PCM-) data stream. The only chance we have is to
+// manually skip this data in the FIQ handler. With sampling rates different
+// from 48000Hz not every frame has valid receive data, so the ratio
+// between pcm data and GPIO status data changes. Our FIQ handler is not
+// able to handle this, hence this driver only works with 48000Hz sampling
+// rate.
+// Reading and writing AC97 registers is another challenge. The core
+// provides us status bits when the read register is updated with *another*
+// value. When we read the same register two times (and the register still
+// contains the same value) these status bits are not set. We work
+// around this by not polling these bits but only wait a fixed delay.
 
 #include <linux/init.h>
 #include <linux/io.h>
@@ -385,8 +380,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 {
 	struct fsl_ssi *ssi = dev_id;
 	struct regmap *regs = ssi->regs;
-	__be32 sisr;
-	__be32 sisr2;
+	u32 sisr, sisr2;
 
 	regmap_read(regs, REG_SSI_SISR, &sisr);
 
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index 18f8dd5209d5..0bdda608d414 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -1,12 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 and i.MX SoC
  *
  * Author: Timur Tabi <timur@freescale.com>
  *
- * Copyright 2007-2008 Freescale Semiconductor, Inc.  This file is licensed
- * under the terms of the GNU General Public License version 2.  This
- * program is licensed "as is" without any warranty of any kind, whether
- * express or implied.
+ * Copyright 2007-2008 Freescale Semiconductor, Inc.
  */
 
 #ifndef _MPC8610_I2S_H
diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c
index 7aac63e2c561..1255dfe19eef 100644
--- a/sound/soc/fsl/fsl_ssi_dbg.c
+++ b/sound/soc/fsl/fsl_ssi_dbg.c
@@ -1,14 +1,10 @@
-/*
- * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions
- *
- * Copyright 2014 Markus Pargmann <mpa@pengutronix.de>, Pengutronix
- *
- * Splitted from fsl_ssi.c
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions
+//
+// Copyright 2014 Markus Pargmann <mpa@pengutronix.de>, Pengutronix
+//
+// Split from fsl_ssi.c
 
 #include <linux/debugfs.h>
 #include <linux/device.h>
@@ -146,7 +142,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev)
 	if (!ssi_dbg->dbg_dir)
 		return -ENOMEM;
 
-	ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO,
+	ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444,
 						 ssi_dbg->dbg_dir, ssi_dbg,
 						 &fsl_ssi_stats_ops);
 	if (!ssi_dbg->dbg_stats) {