Merge branch 'fix/hda' into topic/hda

Speaker-Out renames are merged.

Conflicts:
	sound/pci/hda/patch_realtek.c

12 files changed, 79 insertions, 35 deletions

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6e..496f14c1a731 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
                 err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
                 if (err < 0)
                         goto out_err;
+                opl3->private_data = chip;
         }
 
-        opl3->private_data = chip;
-
         sprintf(card->longname, "%s at 0x%lx, irq %i",
                 card->shortname, chip->ctrl_io, chip->irq);
 
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 65c01798d843..76bac4fc0472 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
         parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
         parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
         parm |= index << AC_AMP_SET_INDEX_SHIFT;
-        parm |= val;
+        if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+            (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+                ; /* set the zero value as a fake mute */
+        else
+                parm |= val;
         snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
         info->vol[ch] = val;
 }
@@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
         val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
         val1 += ofs;
         val1 = ((int)val1) * ((int)val2);
-        if (min_mute)
+        if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
                 val2 |= TLV_DB_SCALE_MUTE;
         if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
                 return -EFAULT;
@@ -5123,7 +5127,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
         const char *pfx = "", *sfx = "";
 
         /* handle as a speaker if it's a fixed line-out */
-        if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT)
+        if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
                 name = "Speaker";
         /* check the location */
         switch (attr) {
@@ -5182,7 +5186,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 
         switch (get_defcfg_device(def_conf)) {
         case AC_JACK_LINE_OUT:
-                return fill_audio_out_name(codec, nid, cfg, "Line-Out",
+                return fill_audio_out_name(codec, nid, cfg, "Line Out",
                                            label, maxlen, indexp);
         case AC_JACK_SPEAKER:
                 return fill_audio_out_name(codec, nid, cfg, "Speaker",
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 654d2e41e25d..9a9f372e1be4 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -298,6 +298,9 @@ enum {
 #define AC_AMPCAP_MUTE                  (1<<31)    /* mute capable */
 #define AC_AMPCAP_MUTE_SHIFT            31
 
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE              (1 << 30) /* min-volume = mute */
+
 /* Connection list */
 #define AC_CLIST_LENGTH                 (0x7f<<0)
 #define AC_CLIST_LONG                   (1<<7)
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bc5a993d1146..c83ccdba1e5a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
                 "Front Speaker", "Surround Speaker", "Bass Speaker"
         };
         static const char * const line_outs[] = {
-                "Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+                "Front Line Out", "Surround Line Out", "Bass Line Out"
         };
 
         fix_volume_caps(codec, dac);
@@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
                 if (num_ctls > 1)
                         name = line_outs[idx];
                 else
-                        name = "Line-Out";
+                        name = "Line Out";
                 break;
         }
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f3b79031fcca..5a56fda83625 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3470,7 +3470,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
                 "Disabled", "Enabled"
         };
         static const char * const texts3[] = {
-                "Disabled", "Speaker Only", "Line-Out+Speaker"
+                "Disabled", "Speaker Only", "Line Out+Speaker"
         };
         const char * const *texts;
 
@@ -4112,7 +4112,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
                 err = snd_hda_ctl_add(codec, nid, kctl);
                 if (err < 0)
                         return err;
-                if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+                if (!(query_amp_caps(codec, nid, hda_dir) &
+                      (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
                         break;
         }
         return 0;
@@ -4413,6 +4414,22 @@ static const struct snd_pci_quirk cxt_fixups[] = {
         {}
 };
 
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+        static hda_nid_t out_nids[] = {
+                0x10, 0x11, 0
+        };
+        hda_nid_t *p;
+
+        for (p = out_nids; *p; p++)
+                snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+                                          AC_AMPCAP_MIN_MUTE |
+                                          query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
 static int patch_conexant_auto(struct hda_codec *codec)
 {
         struct conexant_spec *spec;
@@ -4431,6 +4448,9 @@ static int patch_conexant_auto(struct hda_codec *codec)
         case 0x14f15045:
                 spec->single_adc_amp = 1;
                 break;
+        case 0x14f15051:
+                add_cx5051_fake_mutes(codec);
+                break;
         }
 
         apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e142f6f5c499..01179d53edcd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -841,7 +841,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
                 "Disabled", "Enabled"
         };
         static const char * const texts3[] = {
-                "Disabled", "Speaker Only", "Line-Out+Speaker"
+                "Disabled", "Speaker Only", "Line Out+Speaker"
         };
         const char * const *texts;
 
@@ -1856,7 +1856,7 @@ DEFINE_CAPMIX_NOSRC(3);
  */
 static const char * const alc_slave_pfxs[] = {
         "Front", "Surround", "Center", "LFE", "Side",
-        "Headphone", "Speaker", "Mono", "Line-Out",
+        "Headphone", "Speaker", "Mono", "Line Out",
         "CLFE", "Bass Speaker", "PCM",
         NULL,
 };
@@ -4147,7 +4147,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec)
         else
                 nums = spec->num_adc_nids;
         for (c = 0; c < nums; c++)
-                alc_mux_select(codec, 0, spec->cur_mux[c], true);
+                alc_mux_select(codec, c, spec->cur_mux[c], true);
 }
 
 /* add mic boosts if needed */
@@ -5082,12 +5082,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec,
                                    const struct alc_fixup *fix, int action)
 {
         if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+                /* fake the connections during parsing the tree */
                 hda_nid_t conn1[2] = { 0x0c, 0x0d };
                 hda_nid_t conn2[2] = { 0x0e, 0x0f };
                 snd_hda_override_conn_list(codec, 0x14, 2, conn1);
                 snd_hda_override_conn_list(codec, 0x15, 2, conn1);
                 snd_hda_override_conn_list(codec, 0x18, 2, conn2);
                 snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+        } else if (action == ALC_FIXUP_ACT_PROBE) {
+                /* restore the connections */
+                hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+                snd_hda_override_conn_list(codec, 0x14, 5, conn);
+                snd_hda_override_conn_list(codec, 0x15, 5, conn);
+                snd_hda_override_conn_list(codec, 0x18, 5, conn);
+                snd_hda_override_conn_list(codec, 0x1a, 5, conn);
         }
 }
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8c346ac59d46..5988dbdedc4e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4638,7 +4638,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
                 unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
                 if (no_hp_sensing(spec, i))
                         continue;
-                if (presence)
+                if (1 /*presence*/)
                         stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
 #if 0 /* FIXME */
 /* Resetting the pinctl like below may lead to (a sort of) regressions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e4..278c0a0575f5 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 
         SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
                          0, 0xFF, 1, out_tlv),
-
-        SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
 };
 
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
-        SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+        SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
 
 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
         SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
         SND_SOC_DAPM_OUTPUT("HPOUTR"),
         SND_SOC_DAPM_OUTPUT("LINEOUT"),
 
-        SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
-                           &ak4642_hpout_mixer_controls[0],
-                           ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+        SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+        SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+                            &ak4642_headphone_control),
 
-        SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
-                           &ak4642_hpout_mixer_controls[0],
-                           ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+        SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
 
         SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
                            &ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 static const struct snd_soc_dapm_route ak4642_intercon[] = {
 
         /* Outputs */
-        {"HPOUTL", NULL, "HPOUTL Mixer"},
-        {"HPOUTR", NULL, "HPOUTR Mixer"},
+        {"HPOUTL", NULL, "HPL Out"},
+        {"HPOUTR", NULL, "HPR Out"},
         {"LINEOUT", NULL, "LINEOUT Mixer"},
 
-        {"HPOUTL Mixer", "DACH", "DAC"},
-        {"HPOUTR Mixer", "DACH", "DAC"},
+        {"HPL Out", NULL, "Headphone Enable"},
+        {"HPR Out", NULL, "Headphone Enable"},
+
+        {"Headphone Enable", "Switch", "DACH"},
+
+        {"DACH", NULL, "DAC"},
+
         {"LINEOUT Mixer", "DACL", "DAC"},
 };
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 29c4b02c4790..0ac228b7dc04 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
         return 0;
 }
 
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
 
 static const struct soc_enum str_enum =
         SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 01d1f749cf02..b6adbed6e506 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
                 break;
         case SND_SOC_DAIFMT_DSP_A:
                 /* data on rising edge of bclk, frame high 1clk before data */
-                strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+                strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
                 break;
         }
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1f55ded4047f..1315663c1c09 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
          * standby.
          */
         if (powerdown) {
-                snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+                if (dapm->bias_level == SND_SOC_BIAS_ON)
+                        snd_soc_dapm_set_bias_level(dapm,
+                                                    SND_SOC_BIAS_PREPARE);
                 dapm_seq_run(dapm, &down_list, 0, false);
-                snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+                if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+                        snd_soc_dapm_set_bias_level(dapm,
+                                                    SND_SOC_BIAS_STANDBY);
         }
 }
 
@@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
 
         list_for_each_entry(codec, &card->codec_dev_list, list) {
                 soc_dapm_shutdown_codec(&codec->dapm);
-                snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+                if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+                        snd_soc_dapm_set_bias_level(&codec->dapm,
+                                                    SND_SOC_BIAS_OFF);
         }
 }
 
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 2cf87f5afed4..fde9a7a29cb6 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
 
         spin_lock(&dev->spinlock);
 
-        if (dev->input_panic || dev->output_panic)
+        if (dev->input_panic || dev->output_panic) {
                 ptr = SNDRV_PCM_POS_XRUN;
+                goto unlock;
+        }
 
         if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
                 ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
                 ptr = bytes_to_frames(sub->runtime,
                                         dev->audio_in_buf_pos[index]);
 
+unlock:
         spin_unlock(&dev->spinlock);
         return ptr;
 }