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From:   Connor McAdams <conmanx360@gmail.com>
Cc:     o-takashi@sakamocchi.jp, Connor McAdams <conmanx360@gmail.com>,
        Jaroslav Kysela <perex@perex.cz>,
        Takashi Iwai <tiwai@suse.com>,
        =?UTF-8?q?J=C3=A9r=C3=A9my=20Lefaure?= 
        <jeremy.lefaure@lse.epita.fr>, alsa-devel@alsa-project.org,
        linux-kernel@vger.kernel.org
Subject: [PATCH 7/9] Add DSP Volume set and New mixers for SBZ + R3Di
Date:   Thu,  3 May 2018 17:42:48 -0400
Message-Id: <1525383771-11105-7-git-send-email-conmanx360@gmail.com>
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Adds lookup table for floating point decibel volume, and new functions
to allow for setting the decibel level on the DSP.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 198 ++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 197 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 60b0a14..9d19a275 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -537,6 +537,31 @@ static 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,
@@ -3246,6 +3271,19 @@ 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) }
 
+#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, \
@@ -3258,9 +3296,40 @@ 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
 
@@ -4627,6 +4696,41 @@ 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)
 {
@@ -4728,6 +4832,51 @@ 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)
 {
@@ -4847,6 +4996,39 @@ 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;
@@ -6557,7 +6739,21 @@ static int patch_ca0132(struct hda_codec *codec)
 
 	spec->dsp_state = DSP_DOWNLOAD_INIT;
 	spec->num_mixers = 1;
-	spec->mixers[0] = ca0132_mixer;
+
+	/* 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) {
-- 
2.7.4