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Commit 017310fb authored by Connor McAdams's avatar Connor McAdams Committed by Takashi Iwai
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ALSA: hda/ca0132: Add DSP Volume set and New mixers for SBZ + R3Di 



Adds lookup table for floating point decibel volume, and new functions
to allow for setting the decibel level on the DSP.

Signed-off-by: default avatarConnor McAdams <conmanx360@gmail.com>
Reviewed-by: default avatarTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
parent 7cb9d94c

sound/pci/hda/patch_ca0132.c

+202 −1
Original line number Diff line number Diff line
@@ -541,6 +541,31 @@ static const struct ca0132_alt_out_set alt_out_presets[] = { 
	}

};



/*

 * DSP volume setting structs. Req 1 is left volume, req 2 is right volume,

 * and I don't know what the third req is, but it's always zero. I assume it's

 * some sort of update or set command to tell the DSP there's new volume info.

 */

#define DSP_VOL_OUT 0

#define DSP_VOL_IN  1



struct ct_dsp_volume_ctl {

	hda_nid_t vnid;

	int mid; /* module ID*/

	unsigned int reqs[3]; /* scp req ID */

};



static struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {

	{ .vnid = VNID_SPK,

	  .mid = 0x32,

	  .reqs = {3, 4, 2}

	},

	{ .vnid = VNID_MIC,

	  .mid = 0x37,

	  .reqs = {2, 3, 1}

	}

};



enum hda_cmd_vendor_io {

	/* for DspIO node */

	VENDOR_DSPIO_SCP_WRITE_DATA_LOW      = 0x000,
@@ -3252,6 +3277,24 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, 
	  .tlv = { .c = ca0132_volume_tlv }, \

	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }



/*

 * Creates a mixer control that uses defaults of HDA_CODEC_VOL except for the

 * volume put, which is used for setting the DSP volume. This was done because

 * the ca0132 functions were taking too much time and causing lag.

 */

#define CA0132_ALT_CODEC_VOL_MONO(xname, nid, channel, dir) \

	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \

	  .name = xname, \

	  .subdevice = HDA_SUBDEV_AMP_FLAG, \

	  .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \

			SNDRV_CTL_ELEM_ACCESS_TLV_READ | \

			SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \

	  .info = snd_hda_mixer_amp_volume_info, \

	  .get = snd_hda_mixer_amp_volume_get, \

	  .put = ca0132_alt_volume_put, \

	  .tlv = { .c = snd_hda_mixer_amp_tlv }, \

	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }



#define CA0132_CODEC_MUTE_MONO(xname, nid, channel, dir) \

	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \

	  .name = xname, \
@@ -3264,9 +3307,40 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, 
/* stereo */

#define CA0132_CODEC_VOL(xname, nid, dir) \

	CA0132_CODEC_VOL_MONO(xname, nid, 3, dir)

#define CA0132_ALT_CODEC_VOL(xname, nid, dir) \

	CA0132_ALT_CODEC_VOL_MONO(xname, nid, 3, dir)

#define CA0132_CODEC_MUTE(xname, nid, dir) \

	CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir)



/* lookup tables */

/*

 * Lookup table with decibel values for the DSP. When volume is changed in

 * Windows, the DSP is also sent the dB value in floating point. In Windows,

 * these values have decimal points, probably because the Windows driver

 * actually uses floating point. We can't here, so I made a lookup table of

 * values -90 to 9. -90 is the lowest decibel value for both the ADC's and the

 * DAC's, and 9 is the maximum.

 */

static const unsigned int float_vol_db_lookup[] = {

0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000,

0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000,

0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000,

0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000,

0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000,

0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000,

0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000,

0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000,

0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000,

0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000,

0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000,

0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000,

0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000,

0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000,

0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000,

0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000,

0x40C00000, 0x40E00000, 0x41000000, 0x41100000

};



/* The following are for tuning of products */

#ifdef ENABLE_TUNING_CONTROLS
@@ -4633,6 +4707,41 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, 
/*

 * Volume related

 */

/*

 * Sets the internal DSP decibel level to match the DAC for output, and the

 * ADC for input. Currently only the SBZ sets dsp capture volume level, and

 * all alternative codecs set DSP playback volume.

 */

static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid)

{

	struct ca0132_spec *spec = codec->spec;

	unsigned int dsp_dir;

	unsigned int lookup_val;



	if (nid == VNID_SPK)

		dsp_dir = DSP_VOL_OUT;

	else

		dsp_dir = DSP_VOL_IN;



	lookup_val = spec->vnode_lvol[nid - VNODE_START_NID];



	dspio_set_uint_param(codec,

		ca0132_alt_vol_ctls[dsp_dir].mid,

		ca0132_alt_vol_ctls[dsp_dir].reqs[0],

		float_vol_db_lookup[lookup_val]);



	lookup_val = spec->vnode_rvol[nid - VNODE_START_NID];



	dspio_set_uint_param(codec,

		ca0132_alt_vol_ctls[dsp_dir].mid,

		ca0132_alt_vol_ctls[dsp_dir].reqs[1],

		float_vol_db_lookup[lookup_val]);



	dspio_set_uint_param(codec,

		ca0132_alt_vol_ctls[dsp_dir].mid,

		ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_ZERO);

}



static int ca0132_volume_info(struct snd_kcontrol *kcontrol,

			      struct snd_ctl_elem_info *uinfo)

{
@@ -4734,6 +4843,51 @@ static int ca0132_volume_put(struct snd_kcontrol *kcontrol, 
	return changed;

}



/*

 * This function is the same as the one above, because using an if statement

 * inside of the above volume control for the DSP volume would cause too much

 * lag. This is a lot more smooth.

 */

static int ca0132_alt_volume_put(struct snd_kcontrol *kcontrol,

				struct snd_ctl_elem_value *ucontrol)

{

	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);

	struct ca0132_spec *spec = codec->spec;

	hda_nid_t nid = get_amp_nid(kcontrol);

	int ch = get_amp_channels(kcontrol);

	long *valp = ucontrol->value.integer.value;

	hda_nid_t vnid = 0;

	int changed = 1;



	switch (nid) {

	case 0x02:

		vnid = VNID_SPK;

		break;

	case 0x07:

		vnid = VNID_MIC;

		break;

	}



	/* store the left and right volume */

	if (ch & 1) {

		spec->vnode_lvol[vnid - VNODE_START_NID] = *valp;

		valp++;

	}

	if (ch & 2) {

		spec->vnode_rvol[vnid - VNODE_START_NID] = *valp;

		valp++;

	}



	snd_hda_power_up(codec);

	ca0132_alt_dsp_volume_put(codec, vnid);

	mutex_lock(&codec->control_mutex);

	changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);

	mutex_unlock(&codec->control_mutex);

	snd_hda_power_down(codec);



	return changed;

}



static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag,

			     unsigned int size, unsigned int __user *tlv)

{
@@ -4853,6 +5007,39 @@ static struct snd_kcontrol_new ca0132_mixer[] = { 
	{ } /* end */

};



/*

 * SBZ specific control mixer. Removes auto-detect for mic, and adds surround

 * controls. Also sets both the Front Playback and Capture Volume controls to

 * alt so they set the DSP's decibel level.

 */

static struct snd_kcontrol_new sbz_mixer[] = {

	CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),

	CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),

	CA0132_ALT_CODEC_VOL("Capture Volume", 0x07, HDA_INPUT),

	CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),

	HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),

	HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),

	CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",

				VNID_HP_ASEL, 1, HDA_OUTPUT),

	{ } /* end */

};



/*

 * Same as the Sound Blaster Z, except doesn't use the alt volume for capture

 * because it doesn't set decibel levels for the DSP for capture.

 */

static struct snd_kcontrol_new r3di_mixer[] = {

	CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),

	CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),

	CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT),

	CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),

	HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),

	HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),

	CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",

				VNID_HP_ASEL, 1, HDA_OUTPUT),

	{ } /* end */

};



static int ca0132_build_controls(struct hda_codec *codec)

{

	struct ca0132_spec *spec = codec->spec;
@@ -6566,7 +6753,21 @@ static int patch_ca0132(struct hda_codec *codec) 


	spec->dsp_state = DSP_DOWNLOAD_INIT;

	spec->num_mixers = 1;



	/* Set which mixers each quirk uses. */

	switch (spec->quirk) {

	case QUIRK_SBZ:

		spec->mixers[0] = sbz_mixer;

		snd_hda_codec_set_name(codec, "Sound Blaster Z");

		break;

	case QUIRK_R3DI:

		spec->mixers[0] = r3di_mixer;

		snd_hda_codec_set_name(codec, "Recon3Di");

		break;

	default:

		spec->mixers[0] = ca0132_mixer;

		break;

	}



	/* Setup whether or not to use alt functions */

	switch (spec->quirk) {