Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000' and 'asoc/topic/sh' into asoc-next
Mark Brown
commit
cf1f9e5d16
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@ -5,11 +5,6 @@ Required properties:
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- compatible - "samsung,odroidxu3-audio" - for Odroid XU3 board,
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"samsung,odroidxu4-audio" - for Odroid XU4 board
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- model - the user-visible name of this sound complex
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- 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
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controller
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- 'codec' subnode with a 'sound-dai' property containing list of phandles
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to the CODEC nodes, first entry must be corresponding to the MAX98090
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CODEC and the second entry must be the phandle of the HDMI IP block node
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- clocks - should contain entries matching clock names in the clock-names
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property
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- clock-names - should contain following entries:
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@ -32,12 +27,18 @@ Required properties:
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For Odroid XU4:
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no entries
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Required sub-nodes:
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- 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
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controller
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- 'codec' subnode with a 'sound-dai' property containing list of phandles
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to the CODEC nodes, first entry must be corresponding to the MAX98090
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CODEC and the second entry must be the phandle of the HDMI IP block node
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Example:
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sound {
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compatible = "samsung,odroidxu3-audio";
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samsung,cpu-dai = <&i2s0>;
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samsung,codec-dai = <&max98090>;
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model = "Odroid-XU3";
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samsung,audio-routing =
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"Headphone Jack", "HPL",
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@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
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{ SGTL5000_DAP_AVC_DECAY, 0x0050 },
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};
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/* AVC: Threshold dB -> register: pre-calculated values */
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static const u16 avc_thr_db2reg[97] = {
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0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068,
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0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F,
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0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414,
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0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172,
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0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083,
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0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E,
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0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010,
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0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005,
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0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002,
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0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000,
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0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000};
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/* regulator supplies for sgtl5000, VDDD is an optional external supply */
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enum sgtl5000_regulator_supplies {
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VDDA,
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@ -382,6 +396,65 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol,
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return 0;
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}
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/*
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* custom function to get AVC threshold
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*
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* The threshold dB is calculated by rearranging the calculation from the
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* avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==>
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* dB = ( fls(register_value) - 14.347 ) * 6.02
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*
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* As this calculation is expensive and the threshold dB values may not exeed
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* 0 to 96 we use pre-calculated values.
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*/
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static int avc_get_threshold(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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int db, i;
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u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD);
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/* register value 0 => -96dB */
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if (!reg) {
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ucontrol->value.integer.value[0] = 96;
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ucontrol->value.integer.value[1] = 96;
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return 0;
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}
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/* get dB from register value (rounded down) */
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for (i = 0; avc_thr_db2reg[i] > reg; i++)
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;
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db = i;
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ucontrol->value.integer.value[0] = db;
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ucontrol->value.integer.value[1] = db;
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return 0;
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}
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/*
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* custom function to put AVC threshold
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*
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* The register value is calculated by following formula:
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* register_value = 10^(dB/20) * 0.636 * 2^15
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* As this calculation is expensive and the threshold dB values may not exeed
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* 0 to 96 we use pre-calculated values.
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*/
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static int avc_put_threshold(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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int db;
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u16 reg;
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db = (int)ucontrol->value.integer.value[0];
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if (db < 0 || db > 96)
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return -EINVAL;
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reg = avc_thr_db2reg[db];
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snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg);
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return 0;
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}
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static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
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/* tlv for mic gain, 0db 20db 30db 40db */
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@ -396,6 +469,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0);
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/* tlv for lineout volume, 31 steps of .5db each */
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static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0);
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/* tlv for dap avc max gain, 0db, 6db, 12db */
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static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0);
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/* tlv for dap avc threshold, */
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static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600);
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static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
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/* SOC_DOUBLE_S8_TLV with invert */
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{
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@ -434,6 +513,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
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0x1f, 1,
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lineout_volume),
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SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1),
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/* Automatic Volume Control (DAP AVC) */
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SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0),
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SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0),
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SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0,
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avc_max_gain),
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SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0),
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SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD,
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0, 96, 0, avc_get_threshold, avc_put_threshold,
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avc_threshold),
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};
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/* mute the codec used by alsa core */
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@ -44,7 +44,7 @@ struct s3c24xx_uda134x {
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static unsigned int rates[33 * 2];
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#ifdef ENFORCE_RATES
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static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
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static const struct snd_pcm_hw_constraint_list hw_constraints_rates = {
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.count = ARRAY_SIZE(rates),
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.list = rates,
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.mask = 0,
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@ -301,7 +301,12 @@ struct fsi_master {
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spinlock_t lock;
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};
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static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io);
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static inline int fsi_stream_is_play(struct fsi_priv *fsi,
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struct fsi_stream *io)
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{
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return &fsi->playback == io;
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}
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/*
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* basic read write function
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@ -489,12 +494,6 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi)
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/*
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* fsi_stream_xx() function
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*/
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static inline int fsi_stream_is_play(struct fsi_priv *fsi,
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struct fsi_stream *io)
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{
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return &fsi->playback == io;
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}
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static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi,
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struct snd_pcm_substream *substream)
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{
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