Merge tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

sound fixes for 3.3-rc6 from Takashi Iwai

This contains again regression fixes for various HD-audio and ASoC
regarding SSI and dapm shutdown path.  In addition, a minor azt3328
fix and the correction of the new jack-notification strings in HD-audio.

* tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Kill hyphenated names
  ALSA: hda - Add a fake mute feature
  ALSA: hda - Always set HP pin in unsol handler for STAC/IDT codecs
  ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3
  ALSA: hda/realtek - Fix resume of multiple input sources
  ASoC: i.MX SSI: Fix DSP_A format.
  ASoC: dapm: Check for bias level when powering down

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6e7412e3cca49ab6eb175d55b0983d..496f14c1a731e78071d30c814f6f0e08c143d85e 100644 (file)
--- 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 c2c65f63bf068a0d39fbfd271be81d74c22d0459..684307372d73e87535323ecc2e250a2ffab21f8e 100644 (file)
--- 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;
@@ -5114,7 +5118,7 @@ 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) {
@@ -5173,7 +5177,7 @@ 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 e9f71dc0d46415587f7461678d5ae6051aa9b289..f0f1943a4b2c890ce4affe17425de6210e53ea89 100644 (file)
--- 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 bc5a993d11461868a2115d5b81efec25114a9996..c83ccdba1e5afc1dca9715a870eedf68a8961ee8 100644 (file)
--- 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 a7a5733aa4d20d2ea25edf104e4568b4e42cab3d..d29d6d37790425924ce818acff4f6402f0fa8e82 100644 (file)
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3482,7 +3482,7 @@ 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;
 
@@ -4079,7 +4079,8 @@ 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;
@@ -4379,6 +4380,22 @@ 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;
@@ -4397,6 +4414,9 @@ 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 3647baa9bfed302b63e6fcaf8f9726da7d050389..f286bb8fda1375317589949af5da9a9605bea134 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -802,7 +802,7 @@ 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 @@ static const char * const alc_slave_vols[] = {
        "Headphone Playback Volume",
        "Speaker Playback Volume",
        "Mono Playback Volume",
-       "Line-Out Playback Volume",
+       "Line Out Playback Volume",
        "CLFE Playback Volume",
        "Bass Speaker Playback Volume",
        "PCM Playback Volume",
@@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = {
        "Speaker Playback Switch",
        "Mono Playback Switch",
        "IEC958 Playback Switch",
-       "Line-Out Playback Switch",
+       "Line Out Playback Switch",
        "CLFE Playback Switch",
        "Bass Speaker Playback Switch",
        "PCM Playback Switch",
@@ -3797,7 +3797,7 @@ 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 */

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df131a005a7202b696ccea02fb5dbfeb0898..9dbb5735d778692c81f0010311b2958309a9fa43 100644 (file)
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4629,7 +4629,7 @@ 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/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 01d1f749cf021908ead5cb27e30966a17f144ca6..b6adbed6e506c1585503dff1355b46d4afbaa4cf 100644 (file)
--- 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 1f55ded4047f03b9a538af971c01018f0fb5df10..1315663c1c0990787f226c8358700bce5e0d7963 100644 (file)
--- 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);
        }
 }