Merge tag 'sound-5.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been some significant changes in the core side, both for
  ALSA and ASoC, while lots of development have been seen in SOF, as
  well as many small fixes/improvements for ASoC codecs and platforms.
  Below is a highlight in this cycle:

  Core:
   - The unification of PCM vmalloc buffer allocation helpers into the
     standard API
   - Clean up of the default PCM mmap handling for vmalloc & SG-buffer
   - Fix potential races at ALSA timer open
   - A few new PCM API extensions; just preliminary core changes, the
     actual changes in drivers will be merged in 5.6
   - Continued ASoC componentization works; now almost everything is a
     common ASoC component object. A lot of refactoring and
     simplification have been done along with it.

  ASoC:
   - Many fixes to the Sound Open Firmware (SOF) code
   - Wake on voice support for Chromebooks
   - SPI support and trigger word detection for RT5677
   - New drivers for Analog Devices ADAU7118, Intel Cannonlake systems
     with RT1011 and RT5682, Texas Instruments TAS2562 and TAS2770

  HD-audio:
   - Improved Intel DSP configuration / probe code for SOF
   - Plumbing the legacy HD-audio driver with Intel SOF HDMI
   - DP-MST support for Nvidia HDMI codecs
   - Realtek quirks cleanups and new additions as usual

  Others:
   - Lots of refactoring and cleanups for FireWire; period-size sharing,
     h/w IRQ interval configuration, clock recovery improvements, etc
   - USB-audio: Scarlett mixer quirks
   - Cleanups of PCM calls in various drivers (including media and USB)
     to adapt the core API changes"

* tag 'sound-5.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (497 commits)
  ALSA: usb-audio: Fix Focusrite Scarlett 6i6 gen1 - input handling
  ALSA: hda/realtek - Enable internal speaker of ASUS UX431FLC
  ALSA: aloop: Fix dependency on timer API
  ASoC: DMI long name - avoid to add board name if matches with product name
  ASoC: improve the DMI long card code in asoc-core
  ASoC: rsnd: fix DALIGN register for SSIU
  ALSA: aloop: Avoid unexpected timer event callback tasklets
  ALSA: aloop: Remove redundant locking in timer open function
  ASoC: component: Add sync_stop PCM ops
  ASoC: pcm: Make ioctl ops optional
  ALSA: hda/hdmi - Clear codec->relaxed_resume flag at unbinding
  ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs
  ALSA: cs4236: fix error return comparison of an unsigned integer
  ALSA: usb-audio: Fix NULL dereference at parsing BADD
  ALSA: usb-audio: Fix Scarlett 6i6 Gen 2 port data
  ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop
  ALSA: hda/realtek - Move some alc236 pintbls to fallback table
  ALSA: hda/realtek - Move some alc256 pintbls to fallback table
  ALSA: docs: Update about the new PCM sync_stop ops
  ALSA: pcm: Add card sync_irq field
  ...

1 files changed, 217 insertions, 0 deletions

diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c
new file mode 100644
index 000000000000..d64602950cbd
--- /dev/null
+++ b/sound/soc/samsung/arndale.c
@@ -0,0 +1,217 @@
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright (c) 2014, Insignal Co., Ltd.
+//
+//  Author: Claude <claude@insginal.co.kr>
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/wm8994.h"
+#include "i2s.h"
+
+static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int rfs, ret;
+	unsigned long rclk;
+
+	rfs = 256;
+
+	rclk = params_rate(params) * rfs;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK,
+					0, SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0,
+					0, SND_SOC_CLOCK_OUT);
+
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk, SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops arndale_rt5631_ops = {
+	.hw_params = arndale_rt5631_hw_params,
+};
+
+static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	unsigned int rfs, rclk;
+
+	/* Ensure AIF1CLK is >= 3 MHz for optimal performance */
+	if (params_width(params) == 24)
+		rfs = 384;
+	else if (params_rate(params) == 8000 || params_rate(params) == 11025)
+		rfs = 512;
+	else
+		rfs = 256;
+
+	rclk = params_rate(params) * rfs;
+
+	/*
+	 * We add 1 to the frequency value to ensure proper EPLL setting
+	 * for each audio sampling rate (see epll_24mhz_tbl in drivers/clk/
+	 * samsung/clk-exynos5250.c for list of available EPLL rates).
+	 * The CODEC uses clk API and the value will be rounded hence the MCLK1
+	 * clock's frequency will still be exact multiple of the sample rate.
+	 */
+	return snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1,
+					rclk + 1, SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops arndale_wm1811_ops = {
+	.hw_params = arndale_wm1811_hw_params,
+};
+
+SND_SOC_DAILINK_DEFS(rt5631_hifi,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "rt5631-aif1")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+static struct snd_soc_dai_link arndale_rt5631_dai[] = {
+	{
+		.name = "RT5631 HiFi",
+		.stream_name = "Primary",
+		.dai_fmt = SND_SOC_DAIFMT_I2S
+			| SND_SOC_DAIFMT_NB_NF
+			| SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &arndale_rt5631_ops,
+		SND_SOC_DAILINK_REG(rt5631_hifi),
+	},
+};
+
+SND_SOC_DAILINK_DEFS(wm1811_hifi,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif1")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+static struct snd_soc_dai_link arndale_wm1811_dai[] = {
+	{
+		.name = "WM1811 HiFi",
+		.stream_name = "Primary",
+		.dai_fmt = SND_SOC_DAIFMT_I2S
+			| SND_SOC_DAIFMT_NB_NF
+			| SND_SOC_DAIFMT_CBM_CFM,
+		.ops = &arndale_wm1811_ops,
+		SND_SOC_DAILINK_REG(wm1811_hifi),
+	},
+};
+
+static struct snd_soc_card arndale_rt5631 = {
+	.name = "Arndale RT5631",
+	.owner = THIS_MODULE,
+	.dai_link = arndale_rt5631_dai,
+	.num_links = ARRAY_SIZE(arndale_rt5631_dai),
+};
+
+static struct snd_soc_card arndale_wm1811 = {
+	.name = "Arndale WM1811",
+	.owner = THIS_MODULE,
+	.dai_link = arndale_wm1811_dai,
+	.num_links = ARRAY_SIZE(arndale_wm1811_dai),
+};
+
+static void arndale_put_of_nodes(struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *dai_link;
+	int i;
+
+	for_each_card_prelinks(card, i, dai_link) {
+		of_node_put(dai_link->cpus->of_node);
+		of_node_put(dai_link->codecs->of_node);
+	}
+}
+
+static int arndale_audio_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct snd_soc_card *card;
+	struct snd_soc_dai_link *dai_link;
+	int ret;
+
+	card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev);
+	card->dev = &pdev->dev;
+	dai_link = card->dai_link;
+
+	dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0);
+	if (!dai_link->cpus->of_node) {
+		dev_err(&pdev->dev,
+			"Property 'samsung,audio-cpu' missing or invalid\n");
+		return -EINVAL;
+	}
+
+	if (!dai_link->platforms->name)
+		dai_link->platforms->of_node = dai_link->cpus->of_node;
+
+	dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0);
+	if (!dai_link->codecs->of_node) {
+		dev_err(&pdev->dev,
+			"Property 'samsung,audio-codec' missing or invalid\n");
+		ret = -EINVAL;
+		goto err_put_of_nodes;
+	}
+
+	ret = devm_snd_soc_register_card(card->dev, card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret);
+		goto err_put_of_nodes;
+	}
+	return 0;
+
+err_put_of_nodes:
+	arndale_put_of_nodes(card);
+	return ret;
+}
+
+static int arndale_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	arndale_put_of_nodes(card);
+	return 0;
+}
+
+static const struct of_device_id arndale_audio_of_match[] = {
+	{ .compatible = "samsung,arndale-rt5631",  .data = &arndale_rt5631 },
+	{ .compatible = "samsung,arndale-alc5631", .data = &arndale_rt5631 },
+	{ .compatible = "samsung,arndale-wm1811",  .data = &arndale_wm1811 },
+	{},
+};
+MODULE_DEVICE_TABLE(of, arndale_audio_of_match);
+
+static struct platform_driver arndale_audio_driver = {
+	.driver = {
+		.name = "arndale-audio",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = arndale_audio_of_match,
+	},
+	.probe = arndale_audio_probe,
+	.remove = arndale_audio_remove,
+};
+
+module_platform_driver(arndale_audio_driver);
+
+MODULE_AUTHOR("Claude <claude@insignal.co.kr>");
+MODULE_DESCRIPTION("ALSA SoC Driver for Arndale Board");
+MODULE_LICENSE("GPL");