lkml.org 
[lkml]   [2020]   [Aug]   [21]   [last100]   RSS Feed
Views: [wrap][no wrap]   [headers]  [forward] 
 
Messages in this thread
Patch in this message
/
From
Subject[PATCH 02/20] ALSA: hda/ca0132 - Add speaker tuning initialization commands.
Date
Add speaker tuning initialization DSP commands, and also define
previously unknown DSP command values.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
sound/pci/hda/patch_ca0132.c | 119 +++++++++++++++++++++++++++++++++++
1 file changed, 119 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 7491e2044638..2e664aeee1c4 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -589,6 +589,60 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
}
};

+/*
+ * Definitions for the DSP req's to handle speaker tuning. These all belong to
+ * module ID 0x96, the output effects module.
+ */
+enum speaker_tuning_reqs {
+ /*
+ * Currently, this value is always set to 0.0f. However, on Windows,
+ * when selecting certain headphone profiles on the new Sound Blaster
+ * connect software, the QUERY_SPEAKER_EQ_ADDRESS req on mid 0x80 is
+ * sent. This gets the speaker EQ address area, which is then used to
+ * send over (presumably) an equalizer profile for the specific
+ * headphone setup. It is sent using the same method the DSP
+ * firmware is uploaded with, which I believe is why the 'ctspeq.bin'
+ * file exists in linux firmware tree but goes unused. It would also
+ * explain why the QUERY_SPEAKER_EQ_ADDRESS req is defined but unused.
+ * Once this profile is sent over, SPEAKER_TUNING_USE_SPEAKER_EQ is
+ * set to 1.0f.
+ */
+ SPEAKER_TUNING_USE_SPEAKER_EQ = 0x1f,
+ SPEAKER_TUNING_ENABLE_CENTER_EQ = 0x20,
+ SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL = 0x21,
+ SPEAKER_TUNING_FRONT_RIGHT_VOL_LEVEL = 0x22,
+ SPEAKER_TUNING_CENTER_VOL_LEVEL = 0x23,
+ SPEAKER_TUNING_LFE_VOL_LEVEL = 0x24,
+ SPEAKER_TUNING_REAR_LEFT_VOL_LEVEL = 0x25,
+ SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL = 0x26,
+ SPEAKER_TUNING_SURROUND_LEFT_VOL_LEVEL = 0x27,
+ SPEAKER_TUNING_SURROUND_RIGHT_VOL_LEVEL = 0x28,
+ /*
+ * Inversion is used when setting headphone virtualization to line
+ * out. Not sure why this is, but it's the only place it's ever used.
+ */
+ SPEAKER_TUNING_FRONT_LEFT_INVERT = 0x29,
+ SPEAKER_TUNING_FRONT_RIGHT_INVERT = 0x2a,
+ SPEAKER_TUNING_CENTER_INVERT = 0x2b,
+ SPEAKER_TUNING_LFE_INVERT = 0x2c,
+ SPEAKER_TUNING_REAR_LEFT_INVERT = 0x2d,
+ SPEAKER_TUNING_REAR_RIGHT_INVERT = 0x2e,
+ SPEAKER_TUNING_SURROUND_LEFT_INVERT = 0x2f,
+ SPEAKER_TUNING_SURROUND_RIGHT_INVERT = 0x30,
+ /* Delay is used when setting surround speaker distance in Windows. */
+ SPEAKER_TUNING_FRONT_LEFT_DELAY = 0x31,
+ SPEAKER_TUNING_FRONT_RIGHT_DELAY = 0x32,
+ SPEAKER_TUNING_CENTER_DELAY = 0x33,
+ SPEAKER_TUNING_LFE_DELAY = 0x34,
+ SPEAKER_TUNING_REAR_LEFT_DELAY = 0x35,
+ SPEAKER_TUNING_REAR_RIGHT_DELAY = 0x36,
+ SPEAKER_TUNING_SURROUND_LEFT_DELAY = 0x37,
+ SPEAKER_TUNING_SURROUND_RIGHT_DELAY = 0x38,
+ /* Of these two, only mute seems to ever be used. */
+ SPEAKER_TUNING_MAIN_VOLUME = 0x39,
+ SPEAKER_TUNING_MUTE = 0x3a,
+};
+
/* DSP command sequences for ca0132_alt_select_out */
#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */
struct ca0132_alt_out_set {
@@ -6874,6 +6928,67 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
}
}

+/*
+ * Default speaker tuning values setup for alternative codecs.
+ */
+static const unsigned int sbz_default_delay_values[] = {
+ /* Non-zero values are floating point 0.000198. */
+ 0x394f9e38, 0x394f9e38, 0x00000000, 0x00000000, 0x00000000, 0x00000000
+};
+
+static const unsigned int zxr_default_delay_values[] = {
+ /* Non-zero values are floating point 0.000220. */
+ 0x00000000, 0x00000000, 0x3966afcd, 0x3966afcd, 0x3966afcd, 0x3966afcd
+};
+
+static const unsigned int ae5_default_delay_values[] = {
+ /* Non-zero values are floating point 0.000100. */
+ 0x00000000, 0x00000000, 0x38d1b717, 0x38d1b717, 0x38d1b717, 0x38d1b717
+};
+
+/*
+ * If we never change these, probably only need them on initialization.
+ */
+static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int i, tmp, start_req, end_req;
+ const unsigned int *values;
+
+ switch (ca0132_quirk(spec)) {
+ case QUIRK_SBZ:
+ values = sbz_default_delay_values;
+ break;
+ case QUIRK_ZXR:
+ values = zxr_default_delay_values;
+ break;
+ case QUIRK_AE5:
+ values = ae5_default_delay_values;
+ break;
+ default:
+ values = sbz_default_delay_values;
+ break;
+ }
+
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_ENABLE_CENTER_EQ, tmp);
+
+ start_req = SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL;
+ end_req = SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL;
+ for (i = start_req; i < end_req + 1; i++)
+ dspio_set_uint_param(codec, 0x96, i, tmp);
+
+ start_req = SPEAKER_TUNING_FRONT_LEFT_INVERT;
+ end_req = SPEAKER_TUNING_REAR_RIGHT_INVERT;
+ for (i = start_req; i < end_req + 1; i++)
+ dspio_set_uint_param(codec, 0x96, i, tmp);
+
+
+ for (i = 0; i < 6; i++)
+ dspio_set_uint_param(codec, 0x96,
+ SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]);
+}
+
/*
* Creates a dummy stream to bind the output to. This seems to have to be done
* after changing the main outputs source and destination streams.
@@ -7373,6 +7488,8 @@ static void sbz_setup_defaults(struct hda_codec *codec)
}
}

+ ca0132_alt_init_speaker_tuning(codec);
+
ca0132_alt_create_dummy_stream(codec);
}

@@ -7440,6 +7557,8 @@ static void ae5_setup_defaults(struct hda_codec *codec)
}
}

+ ca0132_alt_init_speaker_tuning(codec);
+
ca0132_alt_create_dummy_stream(codec);
}

--
2.20.1
\
 
 \ /
  Last update: 2020-08-21 20:54    [W:0.122 / U:2.864 seconds]
©2003-2020 Jasper Spaans|hosted at Digital Ocean and TransIP|Read the blog|Advertise on this site