From 896e361e82423aed4490f485dc25de1958c724ed Mon Sep 17 00:00:00 2001 From: Connor McAdams Date: Tue, 25 Aug 2020 16:10:21 -0400 Subject: ALSA: hda/ca0132 - Add speaker tuning initialization commands. Add speaker tuning initialization DSP commands, and also define previously unknown DSP command values. Signed-off-by: Connor McAdams Link: https://lore.kernel.org/r/20200825201040.30339-3-conmanx360@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 119 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 119 insertions(+) (limited to 'sound') 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); } -- cgit v1.2.3