Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...

1 files changed, 25 insertions, 7 deletions

diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index a4c16fd70f77..3eeada57e87d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = {
        2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
 };
 
+static const unsigned int max98088_hp_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0),
+	7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0),
+	15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0),
+	22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0),
+	28, 31, TLV_DB_SCALE_ITEM(150, 50, 0),
+};
+
+static const unsigned int max98088_spk_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0),
+	7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0),
+	15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0),
+	22, 27, TLV_DB_SCALE_ITEM(100, 100, 0),
+	28, 31, TLV_DB_SCALE_ITEM(650, 50, 0),
+};
+
 static const struct snd_kcontrol_new max98088_snd_controls[] = {
 
-       SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L,
-               M98088_REG_3A_LVL_HP_R, 0, 31, 0),
-       SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L,
-               M98088_REG_3E_LVL_SPK_R, 0, 31, 0),
-       SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L,
-               M98088_REG_3C_LVL_REC_R, 0, 31, 0),
+	SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L,
+			 M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv),
+	SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L,
+			 M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv),
+	SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L,
+			 M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv),
 
        SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L,
                M98088_REG_3A_LVL_HP_R, 7, 1, 1),
@@ -2006,7 +2024,7 @@ static int max98088_probe(struct snd_soc_codec *codec)
                        ret);
                goto err_access;
        }
-       dev_info(codec->dev, "revision %c\n", ret + 'A');
+       dev_info(codec->dev, "revision %c\n", ret - 0x40 + 'A');
 
        snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV);