At Thu, 24 Jan 2008 11:54:02 -0500, Matthew Ranostay wrote: > > Some 92HD7xxx codecs have amps on the ports to volume control and/or mute certain ports. > Also this makes stac92hd71bxx unmute amps lines in the init not needed. > > Signed-off-by: Matthew Ranostay <mranostay@xxxxxxxxxxxxxxxxx> The patch will create more controls such as "Headphone Gain Playback Control". What would be a benifit to have both "Headphone" and "Headphone Gain" controls? I'd like to avoid redundancy if both provide a similar functionality. Also, "Line In As Output Gain Playback Volume" is hard to understand. Let's make it simple. thanks, Takashi > --- > diff -r 5bf4c5d02f4b pci/hda/patch_sigmatel.c > --- a/pci/hda/patch_sigmatel.c Thu Jan 24 15:32:15 2008 +0100 > +++ b/pci/hda/patch_sigmatel.c Thu Jan 24 11:25:36 2008 -0500 > @@ -577,10 +577,6 @@ static struct hda_verb stac92hd71bxx_cor > /* connect headphone jack to dac1 */ > { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, > { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ > - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ > - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > }; > > static struct hda_verb stac92hd71bxx_analog_core_init[] = { > @@ -594,11 +590,6 @@ static struct hda_verb stac92hd71bxx_ana > { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ > /* unmute dac0 input in audio mixer */ > { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, > - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ > - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - {} > }; > > static struct hda_verb stac925x_core_init[] = { > @@ -2215,6 +2206,37 @@ static int create_controls(struct sigmat > return 0; > } > > +/* add playback controls for ports that have amps */ > +static int stac92xx_create_amp_ctls(struct hda_codec *codec, > + hda_nid_t nid, char *pfx, int idx) > +{ > + struct sigmatel_spec *spec = codec->spec; > + int err; > + char name[48]; > + u32 caps = query_amp_caps(codec, nid, HDA_INPUT); > + if (idx) > + sprintf(name, "%s %d", pfx, idx); > + else > + strcpy(name, pfx); > + > + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { > + sprintf(name, "%s Playback Volume", name); > + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, > + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); > + if (err < 0) > + return err; > + } > + > + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { > + sprintf(name, "%s Playback Switch", name); > + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, > + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); > + if (err < 0) > + return err; > + } > + return 0; > +} > + > /* add playback controls from the parsed DAC table */ > static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, > const struct auto_pin_cfg *cfg) > @@ -2262,13 +2284,39 @@ static int stac92xx_auto_create_multi_ou > } > } > > - if (spec->line_switch) > - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) > + if (spec->line_switch) { > + int val = cfg->input_pins[AUTO_PIN_LINE] << 8; > + wid_caps = get_wcaps(codec, val >> 8); > + > + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, > + "Line In as Output Switch", val); > + if (err < 0) > return err; > > - if (spec->mic_switch) > - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) > + if (wid_caps & AC_WCAP_IN_AMP) { > + err = stac92xx_create_amp_ctls(codec, val >> 8, > + "Line In as Output Gain", 0); > + if (err < 0) > + return err; > + } > + } > + > + if (spec->mic_switch) { > + int val = cfg->input_pins[AUTO_PIN_MIC] << 8; > + wid_caps = get_wcaps(codec, val >> 8); > + > + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, > + "Mic as Output Switch", val | 1); > + if (err < 0) > return err; > + > + if (wid_caps & AC_WCAP_IN_AMP) { > + err = stac92xx_create_amp_ctls(codec, val >> 8, > + "Mic as Output Gain", 0); > + if (err < 0) > + return err; > + } > + } > > return 0; > } > @@ -2311,6 +2359,13 @@ static int stac92xx_auto_create_hp_ctls( > spec->hp_detect = 1; > nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, > AC_VERB_GET_CONNECT_LIST, 0) & 0xff; > + if (wid_caps & AC_WCAP_IN_AMP) { > + err = stac92xx_create_amp_ctls(codec, > + cfg->hp_pins[i], > + "Headphone Gain", i); > + if (err < 0) > + return err; > + } > if (check_in_dac_nids(spec, nid)) > nid = 0; > if (! nid) > @@ -2320,6 +2375,14 @@ static int stac92xx_auto_create_hp_ctls( > for (i = 0; i < cfg->speaker_outs; i++) { > nid = snd_hda_codec_read(codec, cfg->speaker_pins[i], 0, > AC_VERB_GET_CONNECT_LIST, 0) & 0xff; > + if (get_wcaps(codec, cfg->speaker_pins[i]) & AC_WCAP_IN_AMP) { > + err = stac92xx_create_amp_ctls(codec, > + cfg->speaker_pins[i], > + "Speaker Gain", i); > + if (err < 0) > + return err; > + } > + > if (check_in_dac_nids(spec, nid)) > nid = 0; > if (! nid) > @@ -2329,6 +2392,13 @@ static int stac92xx_auto_create_hp_ctls( > for (i = 0; i < cfg->line_outs; i++) { > nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, > AC_VERB_GET_CONNECT_LIST, 0) & 0xff; > + if (get_wcaps(codec, cfg->line_out_pins[i]) & AC_WCAP_IN_AMP) { > + err = stac92xx_create_amp_ctls(codec, > + cfg->line_out_pins[i], > + "Line Out Gain", i); > + if (err < 0) > + return err; > + } > if (check_in_dac_nids(spec, nid)) > nid = 0; > if (! nid) > _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel