March 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: VIA HDA: Add VT1708S power management function
by Lydia Wang 23 Mar '11

23 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add VT1708S power management function. Define set_widgets_power_state() function for VT1708S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3785,6 +3785,7 @@ spec->stream_name_analog = "VT1818S Analog"; spec->stream_name_digital = "VT1818S Digital"; } + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; }

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Re: [alsa-devel] 答复: [PATCH] ALSA: VIA HDA: Add new power management function
by Takashi Iwai 23 Mar '11

23 Mar '11

At Wed, 23 Mar 2011 15:52:04 +0800, <LydiaWang(a)viatech.com.cn> wrote: > > > > -----邮件原件----- > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > 发送时间: 2011年3月23日 15:49 > > 收件人: Lydia Wang > > 抄送: alsa-devel(a)alsa-project.org; Harald Welte > > 主题: Re: [PATCH] ALSA: VIA HDA: Add new power management function > > > > At Wed, 23 Mar 2011 15:13:28 +0800, > > Lydia Wang wrote: > > > > > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > > > Subject: ALSA: VIA HDA: Add new power management function. > > > > > > Use set_widgets_power_state() function to seperately control different > > > codecs' power management actions and to replace the original large > > > function. Also fix some wrong widgets power up sequence which caused > > > no sound issue under Smart5.1 mode and Independent HP mode. > > > > > > Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> > > > > Applied now. Thanks. > > > > > > Takashi > > > Thank you! > Can I continue to submit the rest patches? Sure. Takashi

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[alsa-devel] [PATCH] ALSA: VIA HDA: Add new power management function
by Lydia Wang 23 Mar '11

23 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add new power management function. Use set_widgets_power_state() function to seperately control different codecs' power management actions and to replace the original large function. Also fix some wrong widgets power up sequence which caused no sound issue under Smart5.1 mode and Independent HP mode. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 937 ++++++++++++++++++++++------------------------ 1 file changed, 452 insertions(+), 485 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -154,6 +154,9 @@ struct delayed_work vt1708_hp_work; int vt1708_jack_detectect; int vt1708_hp_present; + + void (*set_widgets_power_state)(struct hda_codec *codec); + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -245,7 +248,6 @@ }; static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); -static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); static void vt1708_start_hp_work(struct via_spec *spec) @@ -271,6 +273,12 @@ cancel_delayed_work_sync(&spec->vt1708_hp_work); } +static void set_widgets_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + if (spec->set_widgets_power_state) + spec->set_widgets_power_state(codec); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -278,7 +286,7 @@ int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { if (is_aa_path_mute(codec)) @@ -602,482 +610,6 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } -static void set_jack_power_state(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - - if (spec->codec_type == VT1702) { - imux_is_smixer = snd_hda_codec_read( - codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x16, &parm); - set_pin_power_state(codec, 0x17, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, - parm); - } else if (spec->codec_type == VT1708B_8CH - || spec->codec_type == VT1708B_4CH - || spec->codec_type == VT1708S) { - /* SW0 (17h) = stereo mixer */ - int is_8ch = spec->codec_type != VT1708B_4CH; - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1718S) { - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW0/1 (24h/25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - set_pin_power_state(codec, 0x25, &parm); - if (!spec->hp_independent_mode) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, - parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - if (spec->hp_independent_mode) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1716S) { - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - snd_hda_codec_write( - codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Smart 5.1 PW5(1eh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->hp_independent_mode && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); - } else if (spec->codec_type == VT2002P) { - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, - 0, AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Class-D */ - /* PW0 (24h), MW0(18h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* Mono Out */ - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - present = snd_hda_jack_detect(codec, 0x26); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else if (spec->codec_type == VT1812) { - unsigned int present; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } -} - /* * input MUX handling */ @@ -1120,7 +652,7 @@ spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return ret; } @@ -1225,7 +757,7 @@ spec->hp_independent_mode); } /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return 0; } @@ -1443,7 +975,7 @@ } } spec->smart51_enabled = *ucontrol->value.integer.value; - set_jack_power_state(codec); + set_widgets_power_state(codec); return 1; } @@ -1967,7 +1499,7 @@ } /* init power states */ - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(codec, 1); via_free_kctls(codec); /* no longer needed */ @@ -2195,7 +1727,7 @@ if (res & VIA_GPIO_EVENT) via_gpio_control(codec); if (res & VIA_JACK_EVENT) - set_jack_power_state(codec); + set_widgets_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); if (res & VIA_SPEAKER_EVENT) @@ -3642,6 +3174,74 @@ { } /* end */ }; #endif + +static void set_widgets_power_state_vt1708B(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + int is_8ch = 0; + if (spec->codec_type != VT1708B_4CH) + is_8ch = 1; + + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0)); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { @@ -3692,6 +3292,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -3742,6 +3344,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -4438,6 +4042,37 @@ }; #endif +static void set_widgets_power_state_vt1702(struct hda_codec *codec) +{ + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x17, &parm); + set_pin_power_state(codec, 0x16, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); +} + static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -4484,6 +4119,7 @@ spec->loopback.amplist = vt1702_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -4825,6 +4461,72 @@ }; #endif +static void set_widgets_power_state_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2b, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW0 (24h), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + + /* PW1 (25h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2a, &parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1718S(struct hda_codec *codec) { struct via_spec *spec; @@ -4886,6 +4588,8 @@ spec->loopback.amplist = vt1718S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1718S; + return 0; } @@ -4925,8 +4629,7 @@ snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_jack_power_state(codec); - + set_widgets_power_state(codec); return 1; } @@ -5285,6 +4988,99 @@ }; #endif +static void set_widgets_power_state_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_CONNECT_SEL, 0x00) == 5); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write(codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_jack_detect(codec, 0x1c); + + if (present) + mono_out = 0; + else { + present = snd_hda_jack_detect(codec, 0x1d); + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); +} + static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -5339,6 +5135,7 @@ spec->loopback.amplist = vt1716S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -5609,6 +5406,83 @@ }; #endif +static void set_widgets_power_state_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Class-D */ + /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono Out */ + present = snd_hda_jack_detect(codec, 0x26); + + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) @@ -5660,6 +5534,7 @@ spec->loopback.amplist = vt2002P_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -5931,6 +5806,97 @@ }; #endif +static void set_widgets_power_state_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_jack_detect(codec, 0x28); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + +} /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) @@ -5984,6 +5950,7 @@ spec->loopback.amplist = vt1812_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; }

2 1

[PATCH] ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
by David Henningsson 23 Mar '11

23 Mar '11

Similar to commit 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4, this patch avoids unnecessary volume control indices for more Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side" controls. These indices cause these volume controls to be ignored by PulseAudio and vmaster and should be removed whenever possible. Cc: stable(a)kernel.org Reported-by: Jan Losinski <losinski(a)wh2.tu-dresden.de> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++-------- 1 files changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1a03f2..dc67c5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16006,9 +16006,12 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc861_create_out_sw(codec, name, nid, i, 3); + index = 0; + } + err = __alc861_create_out_sw(codec, name, nid, index, 3); if (err < 0) return err; } @@ -17159,16 +17162,19 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -19217,12 +19223,15 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc662_add_vol_ctl(spec, name, nid, i, 3); + index = 0; + } + err = __alc662_add_vol_ctl(spec, name, nid, index, 3); if (err < 0) return err; - err = __alc662_add_sw_ctl(spec, name, mix, i, 3); + err = __alc662_add_sw_ctl(spec, name, mix, index, 3); if (err < 0) return err; } -- 1.7.1 --------------010401080105070700060709--

1 0

Re: [alsa-devel] issue with AD1989A HDA codec and alsamixer
by Raymond Yau 23 Mar '11

23 Mar '11

2011/2/7 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> > Hi Raymond, > > Thanks for your reply. > > To reply to your question, indeed the AD1989A codec is seen by alsa as a > device containing three subdevices (the three ADC) : > > result of arecord -l : > List of CAPTURE Hardware Devices : > > card 0: MID [HDA Intel MID], device 0: AD198x Analog [AD198x Analog] > > Subdevices: 3/3 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > Subdevice #2: subdevice #2 > > > To record four channels simultaneously, I have created the following alsa configuration file : > > *# output device > pcm.AD1989A{ > type hw > card 0 > } > > # input device* * > pcm.quad { > type multi > > slaves.a.pcm "hw:0,0,0" # ADC 0 of the codec (left and right mics) > slaves.a.channels 2 > slaves.b.pcm "hw:0,0,1" # ADC 1 of the codec (front and rear mics) > slaves.b.channels 2 > > bindings.0.slave a > bindings.0.channel 0 > bindings.1.slave a > bindings.1.channel 1 > bindings.2.slave b > bindings.2.channel 0 > bindings.3.slave b > bindings.3.channel 1 > } > > # input device containing interleaved data > pcm.quad2 { > type route > slave.pcm "quad" > ttable.0.0 1 > ttable.1.1 1 > ttable.2.2 1 > ttable.3.3 1 > } > > ### CTL DEVICES (controls) > ctl.mixer0 { > type hw > card 0 > }* > > With this configuration file, I am able to record the four channels > simultaneously with the following command : > > arecord -c4 -fS16_LE -r48000 -D «quad2 » -d5 /tmp/recording.wav > > > I found from where came my problem with the capture gains by adding some > logs in the hda_codec.c file. > When I set one of the capture gain, the put_vol_mute function well write in > the AMP_GAIN_MUTE verb of the corresponding ADC selector (0x0c, 0x0d or 0x0e > widget) but the mute bit is always set to 1. > Indeed, the value associated to the AMP_GAIN_MUTE verb is formated as folow > : > bits 0 to 6 : gain > bit 7 : mute > bits 8 to 11 : index > bit 12 : set right amp > bit 13 : set left amp > bit 14 : set input amp > bit 15 : set output amp > > and alsa sets the value of this register from a080 / b080 to a0b6 / b0b6 > when i change the gain from -58.5 dB to 22.5dB > > Do you know how alsa and thus the alsamixer determine the minimum and > maximum values for the capture gain, and how to say to alsa the the gain is > only a 7bits step value ? > > Thanks in advance. > > Best regards > > Vincent > > > > > > 2011/2/5 Raymond Yau <superquad.vortex2(a)gmail.com> > >> 2011/2/3 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> >> >> > Hi Raymond, >> > >> > Many thanks for your answer. >> > >> > Looking the AD1989A datasheet (which can be found here : >> > http://www.conexant.com/products/entry.jsp?id=599) the vendor id of the >> > codec seems to be 11D4989A >> > So, I think the codec taken into account in the "patch_analog.c" file is >> > well the AD1989A because I found these lines in the source code : >> > >> > /* >> > * patch entries >> > */ >> > static struct hda_codec_preset snd_hda_preset_analog[] = { >> > ... >> > { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 }, >> > ... >> > {} /* terminator */ >> > }; >> > >> > Concerning the digital microphone input of the codec, I don't use it. I >> > have four electret microphones connected to the B and C ports of the >> codec >> > (one on the PORT-B_L pin, one on the PORT-B_R pin, one on the PORT-C_L >> pin >> > and one on the PORT-C_R pin). I have connected by software configuration >> the >> > port B to the ADC1 and the port C to the ADC0. >> > Maybe my problem comes from here, because I don't use the codec in a >> > standard configuration ? >> > if it is the case, do you know how can i resolve it ? >> > Do you also know which registers control the three capture gains of the >> > alsamixerl (i thought it was the gains of the ADC0, ADC1 and ADC2 )? >> > >> >> can you post the output of alsa-info.sh >> >> Six 192kHz 92dB ADCs >> - simultaneous record of up to 3 stereo channels >> >> It does not imply you can record 4 channels with the existing hda driver >> which create three subdevices >> > when recording with 4 channels using audacity with my ad1988a, audacity hang after a few seconds or half minute Does snd-hda-inel really suport SNDRV_PCM_INFO_SYNC_START ? ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/314944/316160/311296 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/314945/316161/311296 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=1216/1216/4864, hwptr=1207/316169/317376/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1217/1216/4864, hwptr=1216/316161/317377/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=2432/1216/4864, hwptr=1216/317376/318592/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=2433/1216/4864, hwptr=1216/317377/318593/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=3648/1216/4864, hwptr=1199/318609/319808/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=3649/1216/4864, hwptr=1216/318593/319809/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/319808/321024/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/319809/321025/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=1216/1216/4864, hwptr=1207/321033/322240/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1217/1216/4864, hwptr=1216/321025/322241/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=2432/1216/4864, hwptr=1216/322240/323456/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=2433/1216/4864, hwptr=1216/322241/323457/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=3648/1216/4864, hwptr=1199/323473/324672/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=3649/1216/4864, hwptr=1216/323457/324673/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/324672/325888/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/324673/325889/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=1216/1216/4864, hwptr=1184/325920/327104/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1217/1216/4864, hwptr=1216/325889/327105/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=2432/1216/4864, hwptr=1216/327104/328320/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=2433/1216/4864, hwptr=1216/327105/328321/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=3648/1216/4864, hwptr=1215/328321/329536/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=3649/1216/4864, hwptr=1216/328321/329537/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/329536/330752/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/329537/330753/325888 ALSA pcm_lib.c:172: XRUN: pcmC0D0c:1 [<f895f201>] xrun+0x97/0x9e [snd_pcm] [<f895f859>] snd_pcm_update_state+0x74/0xc1 [snd_pcm] [<f895fe03>] snd_pcm_update_hw_ptr0+0x55d/0x569 [snd_pcm] [<c042823f>] ? default_wake_function+0xb/0xd [<c0421fd3>] ? __wake_up+0x31/0x3b [<f895fe89>] snd_pcm_period_elapsed+0x7a/0xe9 [snd_pcm] [<f8a41744>] azx_interrupt+0x99/0x120 [snd_hda_intel] [<c046548f>] handle_IRQ_event+0x2f/0x64 [<c04664f8>] handle_fasteoi_irq+0x85/0xc0 [<c0466473>] ? handle_fasteoi_irq+0x0/0xc0 [<c0406e6e>] do_IRQ+0xc7/0xfe [<c0405668>] common_interrupt+0x28/0x30 [<c040a112>] ? mwait_idle+0x38/0x4b [<c0403c61>] cpu_idle+0x101/0x134 [<c06a7db4>] start_secondary+0x197/0x19f ======================= ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x9, stream=0x2, channel=0, format=0x11 ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x9, stream=0x2, channel=0, format=0x11 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x8 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x9 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x8 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x9

1 0

[alsa-devel] [PATCH 1/4] ASoC: Support download of WM8958 MBC firmware
by Mark Brown 22 Mar '11

22 Mar '11

Allow userspace to supply an update to the ROM firmware. The firmware request is non-blocking so userspace can load the firmware at its leisure without delaying startup, the driver will begin using the firmware the next time MBC is started after it has been supplied. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 223 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.c | 4 + sound/soc/codecs/wm8994.h | 4 + 3 files changed, 231 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index c4e4baa..88fd4a9 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -30,10 +30,212 @@ #include "wm8994.h" +#define WM_FW_BLOCK_INFO 0xff +#define WM_FW_BLOCK_PM 0x00 +#define WM_FW_BLOCK_X 0x01 +#define WM_FW_BLOCK_Y 0x02 +#define WM_FW_BLOCK_Z 0x03 +#define WM_FW_BLOCK_I 0x06 +#define WM_FW_BLOCK_A 0x08 +#define WM_FW_BLOCK_C 0x0c + +static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, + const struct firmware *fw, bool check) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + u64 data64; + u32 data32; + const u8 *data; + char *str; + size_t block_len, len; + int ret = 0; + + /* Suppress unneeded downloads */ + if (wm8994->cur_fw == fw) + return 0; + + if (fw->size < 32) { + dev_err(codec->dev, "%s: firmware too short\n", name); + goto err; + } + + if (memcmp(fw->data, "WMFW", 4) != 0) { + dev_err(codec->dev, "%s: firmware has bad file magic %08x\n", + name, data32); + goto err; + } + + memcpy(&data32, fw->data + 4, sizeof(data32)); + len = be32_to_cpu(data32); + + memcpy(&data32, fw->data + 8, sizeof(data32)); + data32 = be32_to_cpu(data32); + if ((data32 >> 24) & 0xff) { + dev_err(codec->dev, "%s: unsupported firmware version %d\n", + name, (data32 >> 24) & 0xff); + goto err; + } + if ((data32 & 0xffff) != 8958) { + dev_err(codec->dev, "%s: unsupported target device %d\n", + name, data32 & 0xffff); + goto err; + } + if (((data32 >> 16) & 0xff) != 0xc) { + dev_err(codec->dev, "%s: unsupported target core %d\n", + name, (data32 >> 16) & 0xff); + goto err; + } + + if (check) { + memcpy(&data64, fw->data + 24, sizeof(u64)); + dev_info(codec->dev, "%s timestamp %llx\n", + name, be64_to_cpu(data64)); + } else { + snd_soc_write(codec, 0x102, 0x2); + snd_soc_write(codec, 0x900, 0x2); + } + + data = fw->data + len; + len = fw->size - len; + while (len) { + if (len < 12) { + dev_err(codec->dev, "%s short data block of %d\n", + name, len); + goto err; + } + + memcpy(&data32, data + 4, sizeof(data32)); + block_len = be32_to_cpu(data32); + if (block_len + 8 > len) { + dev_err(codec->dev, "%d byte block longer than file\n", + block_len); + goto err; + } + if (block_len == 0) { + dev_err(codec->dev, "Zero length block\n"); + goto err; + } + + memcpy(&data32, data, sizeof(data32)); + data32 = be32_to_cpu(data32); + + switch ((data32 >> 24) & 0xff) { + case WM_FW_BLOCK_INFO: + /* Informational text */ + if (!check) + break; + + str = kzalloc(block_len + 1, GFP_KERNEL); + if (str) { + memcpy(str, data + 8, block_len); + dev_info(codec->dev, "%s: %s\n", name, str); + kfree(str); + } else { + dev_err(codec->dev, "Out of memory\n"); + } + break; + case WM_FW_BLOCK_PM: + case WM_FW_BLOCK_X: + case WM_FW_BLOCK_Y: + case WM_FW_BLOCK_Z: + case WM_FW_BLOCK_I: + case WM_FW_BLOCK_A: + case WM_FW_BLOCK_C: + dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name, + block_len, (data32 >> 24) & 0xff, + data32 & 0xffffff); + + if (check) + break; + + data32 &= 0xffffff; + + wm8994_bulk_write(codec->control_data, + data32 & 0xffffff, + block_len / 2, + (void *)(data + 8)); + + break; + default: + dev_warn(codec->dev, "%s: unknown block type %d\n", + name, (data32 >> 24) & 0xff); + break; + } + + /* Round up to the next 32 bit word */ + block_len += block_len % 4; + + data += block_len + 8; + len -= block_len + 8; + } + + if (!check) { + dev_dbg(codec->dev, "%s: download done\n", name); + wm8994->cur_fw = fw; + } else { + dev_info(codec->dev, "%s: got firmware\n", name); + } + + goto ok; + +err: + ret = -EINVAL; +ok: + if (!check) { + snd_soc_write(codec, 0x900, 0x0); + snd_soc_write(codec, 0x102, 0x0); + } + + return ret; +} + static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_pdata *pdata = wm8994->pdata; + int i; + + /* If the DSP is already running then noop */ + if (snd_soc_read(codec, WM8958_DSP2_PROGRAM) & WM8958_DSP2_ENA) + return; + + /* If we have MBC firmware download it */ + if (wm8994->mbc) + wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); + + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, WM8958_DSP2_ENA); + + /* If we've got user supplied MBC settings use them */ + if (pdata && pdata->num_mbc_cfgs) { + struct wm8958_mbc_cfg *cfg + = &pdata->mbc_cfgs[wm8994->mbc_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++) + snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1, + cfg->coeff_regs[i]); + + for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++) + snd_soc_write(codec, + i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1, + cfg->cutoff_regs[i]); + } + + /* Run the DSP */ + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_RUNR); + + /* And we're off! */ + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_ENA | + WM8958_MBC_SEL_MASK, + path << WM8958_MBC_SEL_SHIFT | + WM8958_MBC_ENA); +} + +static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5); int ena, reg, aif, i; @@ -76,6 +278,10 @@ static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) & WM8994_AIF2CLK_ENA_MASK)) return; + /* If we have MBC firmware download it */ + if (wm8994->mbc && wm8994->mbc_ena[mbc]) + wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); + /* Switch the clock over to the appropriate AIF */ snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, @@ -242,6 +448,18 @@ WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), WM8958_MBC_SWITCH("AIF2DAC MBC Switch", 2), }; +static void wm8958_mbc_loaded(const struct firmware *fw, void *context) +{ + struct snd_soc_codec *codec = context; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (fw && wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) { + mutex_lock(&codec->mutex); + wm8994->mbc = fw; + mutex_unlock(&codec->mutex); + } +} + void wm8958_dsp2_init(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -251,6 +469,11 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); + /* We don't require firmware and don't want to delay boot */ + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_mbc.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_mbc_loaded); + if (!pdata) return; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bdd1ac7..f622ff6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1922,6 +1922,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); + wm8994->cur_fw = NULL; + pm_runtime_put(codec->dev); } break; @@ -3136,6 +3138,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } + if (wm8994->mbc) + release_firmware(wm8994->mbc); kfree(wm8994->retune_mobile_texts); kfree(wm8994->drc_texts); kfree(wm8994); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 93a6cf1..1aa365b 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -10,6 +10,7 @@ #define _WM8994_H #include <sound/soc.h> +#include <linux/firmware.h> #include "wm_hubs.h" @@ -114,6 +115,9 @@ struct wm8994_priv { unsigned int aif1clk_disable:1; unsigned int aif2clk_disable:1; + + const struct firmware *cur_fw; + const struct firmware *mbc; }; #endif -- 1.7.4.1

2 4

[alsa-devel] [PATCH] ASoC: Support !REGULATOR build for sgtl5000
by Mark Brown 22 Mar '11

22 Mar '11

The regulator is optional depending on board design. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f7217f..ff29380 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_REGULATOR static int ldo_regulator_is_enabled(struct regulator_dev *dev) { struct ldo_regulator *ldo = rdev_get_drvdata(dev); @@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) return 0; } +#else +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + return -EINVAL; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + return 0; +} +#endif /* * set dac bias -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Fix inconsistent spinlock usage
by Peter Ujfalusi 22 Mar '11

22 Mar '11

The lock is used within the interrupt handler. Correct the spinlock usage, and use irqsave/irqrestore flavour of spin_lock/unlock. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 28 ++++++++++++++++------------ 1 files changed, 16 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8..93f38b1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -670,6 +670,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; unsigned int delay; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: @@ -677,10 +678,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); dac33->t_stamp1 = dac33->t_stamp2; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); @@ -692,11 +693,11 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); /* Move back the timestamp with drain time */ dac33->t_stamp1 -= dac33->mode7_us_to_lthr; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(DAC33_MODE7_MARGIN)); @@ -714,13 +715,14 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); @@ -773,10 +775,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) { struct snd_soc_codec *codec = dev; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) @@ -1173,15 +1176,16 @@ static snd_pcm_sframes_t dac33_dai_delay( unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_BYPASS: break; case DAC33_FIFO_MODE1: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; t1 = dac33->t_stamp2; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ @@ -1246,10 +1250,10 @@ static snd_pcm_sframes_t dac33_dai_delay( } break; case DAC33_FIFO_MODE7: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; uthr = dac33->uthr; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ -- 1.7.4.1

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[alsa-devel] [PATCH 1/2 v4] ALSA: Add snd_ctl_replace() to dynamically replace a control
by Dimitris Papastamos 22 Mar '11

22 Mar '11

Add a function to dynamically replace a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or replace existing controls. A separate patch needs to be made to allow ALSA Mixer to render the replaced controls on the fly. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/control.h | 1 + sound/core/control.c | 65 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..1c8b3ee 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_replace(struct snd_card * card, struct snd_kcontrol * kcontrol, bool add_on_replace); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..a8cae8a 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -368,6 +368,71 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_add); /** + * snd_ctl_replace - replace the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to replace + * @add_on_replace: add the control if not already added + * + * Replaces the given control. If the given control does not exist + * and the add_on_replace flag is set, the control is added. If the + * control exists, it is destroyed first. + * + * Returns zero if successful, or a negative error code on failure. + * + * It frees automatically the control which cannot be added or replaced. + */ +int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, + bool add_on_replace) +{ + struct snd_ctl_elem_id id; + unsigned int idx; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) { + ret = -EINVAL; + goto error; + } + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + if (add_on_replace) + goto add; + up_write(&card->controls_rwsem); + ret = -EINVAL; + goto error; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + goto error; + } +add: + if (snd_ctl_find_hole(card, kcontrol->count) < 0) { + up_write(&card->controls_rwsem); + ret = -ENOMEM; + goto error; + } + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + up_write(&card->controls_rwsem); + for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + return 0; + +error: + snd_ctl_free_one(kcontrol); + return ret; +} + +EXPORT_SYMBOL(snd_ctl_replace); + +/** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance * @kcontrol: the control instance to remove -- 1.7.4.1

3 7

[alsa-devel] Compressed Audio Playback/Capture through ALSA framework
by Patrick Lai 22 Mar '11

22 Mar '11

Hi, Is there a precedent for playback/capture compressed audio stream through ALSA playback/capture interface if underlying hardware supports decoder + sink & encoder + source capabilities? Which source file can I find an example? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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