Merge tag 'sound-4.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This development cycle resulted in a fair amount of changes in both
  core and driver sides. The most significant change in ALSA core is
  about PCM. Also the support of of-graph card and the new DAPM widget
  for DSP are noteworthy changes in ASoC core. And there're lots of
  small changes splat over the tree, as you can see in diffstat.

  Below are a few highlights:

  ALSA core:
   - Removal of set_fs() hackery from PCM core stuff, and the code
     reorganization / optimization thereafter
   - Improved support of PCM ack ops, and a new ABI for improved
     control/status mmap handling
   - Lots of constifications in various codes

  ASoC core:
   - The support of of-graph card, which may work as a better generic
     device for a replacement of simple-card
   - New widget types intended mainly for use with DSPs

  ASoC drivers:
   - New drivers for Allwinner V3s SoCs
   - Ensonic ES8316 codec support
   - More Intel SKL and KBL works
   - More device support for Intel SST Atom (mostly for cheap tablets
     and 2-in-1 devices)
   - Support for Rockchip PDM controllers
   - Support for STM32 I2S and S/PDIF controllers
   - Support for ZTE AUD96P22 codecs

  HD-audio:
   - Support of new Realtek codecs (ALC215/ALC285/ALC289), more quirks
     for HP and Dell machines
   - A few more fixes for i915 component binding"

* tag 'sound-4.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (418 commits)
  ALSA: hda - Fix unbalance of i915 module refcount
  ASoC: Intel: Skylake: Remove driver debugfs exit
  ASoC: Intel: Skylake: explicitly add the headers sst-dsp.h
  ALSA: hda/realtek - Remove GPIO_MASK
  ALSA: hda/realtek - Fix typo of pincfg for Dell quirk
  ALSA: pcm: add a documentation for tracepoints
  ALSA: atmel: ac97c: fix error return code in atmel_ac97c_probe()
  ALSA: x86: fix error return code in hdmi_lpe_audio_probe()
  ASoC: Intel: Skylake: Add support to read firmware registers
  ASoC: Intel: Skylake: Add sram address to sst_addr structure
  ASoC: Intel: Skylake: Debugfs facility to dump module config
  ASoC: Intel: Skylake: Add debugfs support
  ASoC: fix semicolon.cocci warnings
  ASoC: rt5645: Add quirk override by module option
  ASoC: rsnd: make arrays path and cmd_case static const
  ASoC: audio-graph-card: add widgets and routing for external amplifier support
  ASoC: audio-graph-card: update bindings for amplifier support
  ASoC: rt5665: calibration should be done before jack detection
  ASoC: rsnd: constify dev_pm_ops structures.
  ASoC: nau8825: change crosstalk-bypass property to bool type
  ...

1 files changed, 89 insertions, 0 deletions

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 5a2702edeb77..8f6814c1eb6b 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_DAP_AVC_DECAY,		0x0050 },
 };
 
+/* AVC: Threshold dB -> register: pre-calculated values */
+static const u16 avc_thr_db2reg[97] = {
+	0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068,
+	0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F,
+	0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414,
+	0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172,
+	0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083,
+	0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E,
+	0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010,
+	0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005,
+	0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002,
+	0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000};
+
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
 enum sgtl5000_regulator_supplies {
 	VDDA,
@@ -382,6 +396,65 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+/*
+ * custom function to get AVC threshold
+ *
+ * The threshold dB is calculated by rearranging the calculation from the
+ * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==>
+ * dB = ( fls(register_value) - 14.347 ) * 6.02
+ *
+ * As this calculation is expensive and the threshold dB values may not exeed
+ * 0 to 96 we use pre-calculated values.
+ */
+static int avc_get_threshold(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int db, i;
+	u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD);
+
+	/* register value 0 => -96dB */
+	if (!reg) {
+		ucontrol->value.integer.value[0] = 96;
+		ucontrol->value.integer.value[1] = 96;
+		return 0;
+	}
+
+	/* get dB from register value (rounded down) */
+	for (i = 0; avc_thr_db2reg[i] > reg; i++)
+		;
+	db = i;
+
+	ucontrol->value.integer.value[0] = db;
+	ucontrol->value.integer.value[1] = db;
+
+	return 0;
+}
+
+/*
+ * custom function to put AVC threshold
+ *
+ * The register value is calculated by following formula:
+ *                                    register_value = 10^(dB/20) * 0.636 * 2^15
+ * As this calculation is expensive and the threshold dB values may not exeed
+ * 0 to 96 we use pre-calculated values.
+ */
+static int avc_put_threshold(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int db;
+	u16 reg;
+
+	db = (int)ucontrol->value.integer.value[0];
+	if (db < 0 || db > 96)
+		return -EINVAL;
+	reg = avc_thr_db2reg[db];
+	snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg);
+
+	return 0;
+}
+
 static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
 
 /* tlv for mic gain, 0db 20db 30db 40db */
@@ -396,6 +469,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0);
 /* tlv for lineout volume, 31 steps of .5db each */
 static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0);
 
+/* tlv for dap avc max gain, 0db, 6db, 12db */
+static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0);
+
+/* tlv for dap avc threshold, */
+static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600);
+
 static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 	/* SOC_DOUBLE_S8_TLV with invert */
 	{
@@ -434,6 +513,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 			0x1f, 1,
 			lineout_volume),
 	SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1),
+
+	/* Automatic Volume Control (DAP AVC) */
+	SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0),
+	SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0),
+	SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0,
+			avc_max_gain),
+	SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0),
+	SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD,
+			0, 96, 0, avc_get_threshold, avc_put_threshold,
+			avc_threshold),
 };
 
 /* mute the codec used by alsa core */