Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It was a busy development cycle at this time, as you can see a wide
  range of changes in diffstat.  There are no big changes but many
  refactoring and improvements.  Here we go some highlights:

  ALSA core:
   - Procfs codes were cleaned up to use seq_file
   - Procfs can be opt out via Kconfig (only for EXPERT)
   - Two types of jack API were unified finally; now both kctl and input
     jack devs are handled via a single function call.

  HD-audio:
   - Continued code restructuring for the future ASoC driver; now HDA
     controller driver is split to a core helper module.
   - Preliminary codes for Skylake audio support in HDA core.
   - Proper i915 gfx power well management for SKL & co
   - Enabled runtime PM as default for Intel HDMI/DP codecs
   - Newer Tegra chip supports
   - More quirks for Dell headsets, Alienware (with CA0132), etc.
   - A couple of DRM ELD helper API functions

  ASoC:
   - Support for loading ASoC topology maps from firmware, intended to
     be used to allow self-describing DSP firmware images to be built
     which can map controls added by the DSP to userspace without the
     kernel needing to know about individual DSP firmwares
   - Lots of refactoring to avoid direct access to snd_soc_codec where
     it's not needed supporting future refactoring
   - Big refactoring, cleanup and enhancement for the Wolfson ADSP
     driver
   - Cleanup series for TI TAS2552 and R-CAR drivers
   - Fixes and improvements on RT56xx codecs
   - Support for TI TAS571x power amplifiers
   - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
   - Support for x86 systems with RT5650 and Qualcomm Storm
   - Support for Mediatek AFE (Audio Front End) unit
   - Other various small fixes to ASoC codec drivers

  Firewire:
   - Enhanced to allow non-blocking streams to use timestamp
     synchronization
   - Improve support for DM1500 and BeBoBv3

  Misc:
   - Cleanup of old pci API functions over all PCI sound drivers
   - Fix long-standing regression of the old powermac i2c setup"

* tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits)
  ALSA: pcm: Fix pcm_class sysfs output
  ALSA: hda-beep: Update authors dead email address
  ASoC: wm_adsp: Move DSP Rate controls into the codec
  ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
  ALSA: hda: provide default bus io ops extended hdac
  ALSA: hda: add hda link cleanup routine
  ALSA: hda: add hdac_ext stream creation and cleanup routines
  ASoC: rsrc-card: remove unused ret
  ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
  ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
  ASoC: mediatek: Add machine driver for MAX98090 codec
  ASoC: mediatek: Add AFE platform driver
  ASoC: rsnd: remove io from rsnd_mod
  ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
  ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
  ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
  ...

1 files changed, 278 insertions, 0 deletions

diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
new file mode 100644
index 000000000000..094055323059
--- /dev/null
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -0,0 +1,278 @@
+/*
+ * mt8173-rt5650-rt5676.c  --  MT8173 machine driver with RT5650/5676 codecs
+ *
+ * Copyright (c) 2015 MediaTek Inc.
+ * Author: Koro Chen <koro.chen@mediatek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5645.h"
+#include "../codecs/rt5677.h"
+
+#define MCLK_FOR_CODECS		12288000
+
+static const struct snd_soc_dapm_widget mt8173_rt5650_rt5676_widgets[] = {
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mt8173_rt5650_rt5676_routes[] = {
+	{"Speaker", NULL, "SPOL"},
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650  */
+	{"Sub DMIC L1", NULL, "Int Mic"}, /* DMIC from 5676 */
+	{"Sub DMIC R1", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Headphone", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650  */
+	{"Headset Mic", NULL, "micbias1"},
+	{"Headset Mic", NULL, "micbias2"},
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"Sub AIF2RX", NULL, "Headset Mic"}, /* IF2 DAC from 5650  */
+};
+
+static const struct snd_kcontrol_new mt8173_rt5650_rt5676_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream,
+					  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int i, ret;
+
+	for (i = 0; i < rtd->num_codecs; i++) {
+		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+
+		/* pll from mclk 12.288M */
+		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
+					  params_rate(params) * 512);
+		if (ret)
+			return ret;
+
+		/* sysclk from pll */
+		ret = snd_soc_dai_set_sysclk(codec_dai, 1,
+					     params_rate(params) * 512,
+					     SND_SOC_CLOCK_IN);
+		if (ret)
+			return ret;
+	}
+	return 0;
+}
+
+static struct snd_soc_ops mt8173_rt5650_rt5676_ops = {
+	.hw_params = mt8173_rt5650_rt5676_hw_params,
+};
+
+static struct snd_soc_jack mt8173_rt5650_rt5676_jack;
+
+static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_card *card = runtime->card;
+	struct snd_soc_codec *codec = runtime->codec_dais[0]->codec;
+	struct snd_soc_codec *codec_sub = runtime->codec_dais[1]->codec;
+	int ret;
+
+	rt5645_sel_asrc_clk_src(codec,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_AD_STEREO_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+	rt5677_sel_asrc_clk_src(codec_sub,
+				RT5677_DA_STEREO_FILTER |
+				RT5677_AD_STEREO1_FILTER,
+				RT5677_CLK_SEL_I2S1_ASRC);
+	rt5677_sel_asrc_clk_src(codec_sub,
+				RT5677_AD_STEREO2_FILTER |
+				RT5677_I2S2_SOURCE,
+				RT5677_CLK_SEL_I2S2_ASRC);
+
+	/* enable jack detection */
+	ret = snd_soc_card_jack_new(card, "Headset Jack",
+				    SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+				    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+				    SND_JACK_BTN_2 | SND_JACK_BTN_3,
+				    &mt8173_rt5650_rt5676_jack, NULL, 0);
+	if (ret) {
+		dev_err(card->dev, "Can't new Headset Jack %d\n", ret);
+		return ret;
+	}
+
+	return rt5645_set_jack_detect(codec,
+				      &mt8173_rt5650_rt5676_jack,
+				      &mt8173_rt5650_rt5676_jack,
+				      &mt8173_rt5650_rt5676_jack);
+}
+
+static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = {
+	{
+		.dai_name = "rt5645-aif1",
+	},
+	{
+		.dai_name = "rt5677-aif1",
+	},
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "rt5650_rt5676 Playback",
+		.stream_name = "rt5650_rt5676 Playback",
+		.cpu_dai_name = "DL1",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "rt5650_rt5676 Capture",
+		.stream_name = "rt5650_rt5676 Capture",
+		.cpu_dai_name = "VUL",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_capture = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		.name = "Codec",
+		.cpu_dai_name = "I2S",
+		.platform_name = "11220000.mt8173-afe-pcm",
+		.no_pcm = 1,
+		.codecs = mt8173_rt5650_rt5676_codecs,
+		.num_codecs = 2,
+		.init = mt8173_rt5650_rt5676_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &mt8173_rt5650_rt5676_ops,
+		.ignore_pmdown_time = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{ /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */
+		.name = "rt5650_rt5676 intercodec",
+		.stream_name = "rt5650_rt5676 intercodec",
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5677-aif2",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBM_CFM,
+	},
+
+};
+
+static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = {
+	{
+		.name_prefix = "Sub",
+	},
+};
+
+static struct snd_soc_card mt8173_rt5650_rt5676_card = {
+	.name = "mtk-rt5650-rt5676",
+	.dai_link = mt8173_rt5650_rt5676_dais,
+	.num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais),
+	.codec_conf = mt8173_rt5650_rt5676_codec_conf,
+	.num_configs = ARRAY_SIZE(mt8173_rt5650_rt5676_codec_conf),
+	.controls = mt8173_rt5650_rt5676_controls,
+	.num_controls = ARRAY_SIZE(mt8173_rt5650_rt5676_controls),
+	.dapm_widgets = mt8173_rt5650_rt5676_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_rt5676_widgets),
+	.dapm_routes = mt8173_rt5650_rt5676_routes,
+	.num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_rt5676_routes),
+};
+
+static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
+	int ret;
+
+	mt8173_rt5650_rt5676_codecs[0].of_node =
+		of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0);
+	if (!mt8173_rt5650_rt5676_codecs[0].of_node) {
+		dev_err(&pdev->dev,
+			"Property 'audio-codec' missing or invalid\n");
+		return -EINVAL;
+	}
+	mt8173_rt5650_rt5676_codecs[1].of_node =
+		of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1);
+	if (!mt8173_rt5650_rt5676_codecs[1].of_node) {
+		dev_err(&pdev->dev,
+			"Property 'audio-codec' missing or invalid\n");
+		return -EINVAL;
+	}
+	mt8173_rt5650_rt5676_codec_conf[0].of_node =
+		mt8173_rt5650_rt5676_codecs[1].of_node;
+
+	mt8173_rt5650_rt5676_dais[3].codec_of_node =
+		mt8173_rt5650_rt5676_codecs[1].of_node;
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
+			__func__, ret);
+	return ret;
+}
+
+static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
+	{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mt8173_rt5650_rt5676_dt_match);
+
+static struct platform_driver mt8173_rt5650_rt5676_driver = {
+	.driver = {
+		   .name = "mtk-rt5650-rt5676",
+		   .owner = THIS_MODULE,
+		   .of_match_table = mt8173_rt5650_rt5676_dt_match,
+#ifdef CONFIG_PM
+		   .pm = &snd_soc_pm_ops,
+#endif
+	},
+	.probe = mt8173_rt5650_rt5676_dev_probe,
+	.remove = mt8173_rt5650_rt5676_dev_remove,
+};
+
+module_platform_driver(mt8173_rt5650_rt5676_driver);
+
+/* Module information */
+MODULE_DESCRIPTION("MT8173 RT5650 and RT5676 SoC machine driver");
+MODULE_AUTHOR("Koro Chen <koro.chen@mediatek.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:mtk-rt5650-rt5676");
+