Merge tag 'sound-3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "A usual pattern of half ASoC and half HD-audio fixes, although
  HD-audio fixups have more volumes, in addition to a couple of trivial
  fixes.  Nothing to worry much is found here.

  For ASoC side: a few fixes for PCM rate constraints calculations,
  regmap byte-order fix, the rest driver specific fixes (atmel, fsl,
  omap, kirkwood, wm codecs).

  For HD-audio: Dell headset and mono out fix, ELD update in polling
  mode, ALC283 Chromebook fixes, a few fixes for old AD codecs and
  MBA2, one regression fix"

* tag 'sound-3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
  ALSA: hda - Fix silent output on MacBook Air 2,1
  ALSA: hda - Fix missing ELD info when using jackpoll_ms parameter
  ALSA: hda/realtek - remove hp_automute_hook from alc283_fixup_chromebook
  ASoC: wm8731: fix dsp mode configuration
  ALSA: hda/realtek - Independent of model for HP
  ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek)
  ALSA: hda - Use always amps for auto-mute on AD1986A codec
  ALSA: hda/analog - Handle inverted EAPD properly in vmaster hook
  ALSA: hda - Another fixup for ASUS laptop with ALC660 codec
  ALSA: atmel: Fix possible array overflow
  ALSA: hda - Fix complete_all() timing in deferred probes
  ALSA: hda - Fix bad EAPD setup for HP machines with AD1984A
  ASoC: core: fix devres parameter in devm_snd_soc_register_card()
  ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare
  ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe()
  ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable
  ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation
  ALSA: hda - Fix silent output on ASUS W7J laptop
  ASoC: core: Use consistent byte ordering in snd_soc_bytes_get
  ALSA: dice: fix array limits in dice_proc_read()
  ...

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 2037c45adfe648e907fa86290f7dfea847989c20..56ebdfca627375b1a7ec7d5ce5606bc124ecdf42 100644 (file)
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -104,7 +104,8 @@ struct device;
        SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
        .kcontrol_news = wcontrols, .num_kcontrols = 1}
 #define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{      .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, \
+{      .id = snd_soc_dapm_mux, .name = wname, \
+       SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
        .kcontrol_news = wcontrols, .num_kcontrols = 1}
 #define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \
 {      .id = snd_soc_dapm_virt_mux, .name = wname, \

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 872d59e35ee23583cc60a2be24afa5ba1bf3c88d..721d8fd45685f769cbc616a6ba0552b85cd5c841 100644 (file)
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -357,7 +357,8 @@ static int set_sample_rates(struct atmel_abdac *dac)
                if (new_rate < 0)
                        break;
                /* make sure we are below the ABDAC clock */
-               if (new_rate <= clk_get_rate(dac->pclk)) {
+               if (index < MAX_NUM_RATES &&
+                   new_rate <= clk_get_rate(dac->pclk)) {
                        dac->rates[index] = new_rate / 256;
                        index++;
                }

diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c
index 57bcd31fcc123c38cd293cab40b52310734698a2..c0aa64941cee0bd927450a6b69314d14f2f02bce 100644 (file)
--- a/sound/firewire/dice.c
+++ b/sound/firewire/dice.c
@@ -1019,7 +1019,7 @@ static void dice_proc_read(struct snd_info_entry *entry,
 
        if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0)
                return;
-       quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx));
+       quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4);
        for (stream = 0; stream < tx_rx_header.number; ++stream) {
                if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 +
                                       stream * tx_rx_header.size,
@@ -1045,7 +1045,7 @@ static void dice_proc_read(struct snd_info_entry *entry,
 
        if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0)
                return;
-       quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx));
+       quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4);
        for (stream = 0; stream < tx_rx_header.number; ++stream) {
                if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 +
                                       stream * tx_rx_header.size,

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c6d230193da6214700e23369df0d2355c301012a..27aa14007cbd400ebcb1f21e5e4036d58e778430 100644 (file)
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3876,7 +3876,8 @@ static int azx_probe(struct pci_dev *pci,
        }
 
        dev++;
-       complete_all(&chip->probe_wait);
+       if (chip->disabled)
+               complete_all(&chip->probe_wait);
        return 0;
 
 out_free:
@@ -3953,10 +3954,10 @@ static int azx_probe_continue(struct azx *chip)
        if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo)
                pm_runtime_put_noidle(&pci->dev);
 
-       return 0;
-
 out_free:
-       chip->init_failed = 1;
+       if (err < 0)
+               chip->init_failed = 1;
+       complete_all(&chip->probe_wait);
        return err;
 }
 

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a83559f4cbd27e455a941adaddc6d0513d9dd81..cac015be3325d9760366434c6e727c472a21dbf5 100644 (file)
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled)
 
        if (!spec->eapd_nid)
                return;
+       if (codec->inv_eapd)
+               enabled = !enabled;
        snd_hda_codec_update_cache(codec, spec->eapd_nid, 0,
                                   AC_VERB_SET_EAPD_BTLENABLE,
                                   enabled ? 0x02 : 0x00);
@@ -359,6 +361,9 @@ static int patch_ad1986a(struct hda_codec *codec)
         */
        spec->gen.multiout.no_share_stream = 1;
 
+       /* AD1986A can't manage the dynamic pin on/off smoothly */
+       spec->gen.auto_mute_via_amp = 1;
+
        snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
                           ad1986a_fixups);
        snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -962,6 +967,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
        switch (action) {
        case HDA_FIXUP_ACT_PRE_PROBE:
                spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+               spec->gen.own_eapd_ctl = 1;
                snd_hda_sequence_write_cache(codec, gpio_init_verbs);
                break;
        case HDA_FIXUP_ACT_PROBE:

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 08407bed093e5a64d4b659f90bf236dead9ec002..c4a66ef6cf6f02e25134ab3f9b8ae1dfef6224ee 100644 (file)
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 
 static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
 
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack)
 {
        struct hdmi_spec *spec = codec->spec;
+       int pin_idx = pin_nid_to_pin_index(spec, jack->nid);
+       if (pin_idx < 0)
+               return;
+
+       if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
+               snd_hda_jack_report_sync(codec);
+}
+
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+{
        int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
-       int pin_nid;
-       int pin_idx;
        struct hda_jack_tbl *jack;
        int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
 
        jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
        if (!jack)
                return;
-       pin_nid = jack->nid;
        jack->jack_dirty = 1;
 
        _snd_printd(SND_PR_VERBOSE,
                "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
-               codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
+               codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
                !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
-       pin_idx = pin_nid_to_pin_index(spec, pin_nid);
-       if (pin_idx < 0)
-               return;
-
-       if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
-               snd_hda_jack_report_sync(codec);
+       jack_callback(codec, jack);
 }
 
 static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -2095,7 +2097,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
                hda_nid_t pin_nid = per_pin->pin_nid;
 
                hdmi_init_pin(codec, pin_nid);
-               snd_hda_jack_detect_enable(codec, pin_nid, pin_nid);
+               snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid,
+                       codec->jackpoll_interval > 0 ? jack_callback : NULL);
        }
        return 0;
 }

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c770bdba6531c8a0d3eaf2f64e2c38d6d8ad758c..c5ea483d755981cdbe6c24c24ba0c98478e4c709 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1780,6 +1780,7 @@ enum {
        ALC889_FIXUP_DAC_ROUTE,
        ALC889_FIXUP_MBP_VREF,
        ALC889_FIXUP_IMAC91_VREF,
+       ALC889_FIXUP_MBA21_VREF,
        ALC882_FIXUP_INV_DMIC,
        ALC882_FIXUP_NO_PRIMARY_HP,
        ALC887_FIXUP_ASUS_BASS,
@@ -1884,17 +1885,13 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec,
        }
 }
 
-/* Set VREF on speaker pins on imac91 */
-static void alc889_fixup_imac91_vref(struct hda_codec *codec,
-                                    const struct hda_fixup *fix, int action)
+static void alc889_fixup_mac_pins(struct hda_codec *codec,
+                                 const hda_nid_t *nids, int num_nids)
 {
        struct alc_spec *spec = codec->spec;
-       static hda_nid_t nids[2] = { 0x18, 0x1a };
        int i;
 
-       if (action != HDA_FIXUP_ACT_INIT)
-               return;
-       for (i = 0; i < ARRAY_SIZE(nids); i++) {
+       for (i = 0; i < num_nids; i++) {
                unsigned int val;
                val = snd_hda_codec_get_pin_target(codec, nids[i]);
                val |= AC_PINCTL_VREF_50;
@@ -1903,6 +1900,26 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
        spec->gen.keep_vref_in_automute = 1;
 }
 
+/* Set VREF on speaker pins on imac91 */
+static void alc889_fixup_imac91_vref(struct hda_codec *codec,
+                                    const struct hda_fixup *fix, int action)
+{
+       static hda_nid_t nids[2] = { 0x18, 0x1a };
+
+       if (action == HDA_FIXUP_ACT_INIT)
+               alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
+}
+
+/* Set VREF on speaker pins on mba21 */
+static void alc889_fixup_mba21_vref(struct hda_codec *codec,
+                                   const struct hda_fixup *fix, int action)
+{
+       static hda_nid_t nids[2] = { 0x18, 0x19 };
+
+       if (action == HDA_FIXUP_ACT_INIT)
+               alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
+}
+
 /* Don't take HP output as primary
  * Strangely, the speaker output doesn't work on Vaio Z and some Vaio
  * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05
@@ -2102,6 +2119,12 @@ static const struct hda_fixup alc882_fixups[] = {
                .chained = true,
                .chain_id = ALC882_FIXUP_GPIO1,
        },
+       [ALC889_FIXUP_MBA21_VREF] = {
+               .type = HDA_FIXUP_FUNC,
+               .v.func = alc889_fixup_mba21_vref,
+               .chained = true,
+               .chain_id = ALC889_FIXUP_MBP_VREF,
+       },
        [ALC882_FIXUP_INV_DMIC] = {
                .type = HDA_FIXUP_FUNC,
                .v.func = alc_fixup_inv_dmic_0x12,
@@ -2172,7 +2195,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
        SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF),
        SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD),
        SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF),
-       SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF),
+       SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF),
        SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF),
        SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF),
        SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO),
@@ -3287,6 +3310,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
                alc_write_coef_idx(codec, 0x18, 0x7388);
                break;
        case 0x10ec0668:
+               alc_write_coef_idx(codec, 0x11, 0x0001);
                alc_write_coef_idx(codec, 0x15, 0x0d60);
                alc_write_coef_idx(codec, 0xc3, 0x0000);
                break;
@@ -3315,6 +3339,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
                alc_write_coef_idx(codec, 0x18, 0x7388);
                break;
        case 0x10ec0668:
+               alc_write_coef_idx(codec, 0x11, 0x0001);
                alc_write_coef_idx(codec, 0x15, 0x0d50);
                alc_write_coef_idx(codec, 0xc3, 0x0000);
                break;
@@ -3600,11 +3625,6 @@ static void alc283_hp_automute_hook(struct hda_codec *codec,
                            vref);
 }
 
-static void alc283_chromebook_caps(struct hda_codec *codec)
-{
-       snd_hda_override_wcaps(codec, 0x03, 0);
-}
-
 static void alc283_fixup_chromebook(struct hda_codec *codec,
                                    const struct hda_fixup *fix, int action)
 {
@@ -3613,9 +3633,26 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
 
        switch (action) {
        case HDA_FIXUP_ACT_PRE_PROBE:
-               alc283_chromebook_caps(codec);
+               snd_hda_override_wcaps(codec, 0x03, 0);
                /* Disable AA-loopback as it causes white noise */
                spec->gen.mixer_nid = 0;
+               break;
+       case HDA_FIXUP_ACT_INIT:
+               /* Enable Line1 input control by verb */
+               val = alc_read_coef_idx(codec, 0x1a);
+               alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
+               break;
+       }
+}
+
+static void alc283_fixup_sense_combo_jack(struct hda_codec *codec,
+                                   const struct hda_fixup *fix, int action)
+{
+       struct alc_spec *spec = codec->spec;
+       int val;
+
+       switch (action) {
+       case HDA_FIXUP_ACT_PRE_PROBE:
                spec->gen.hp_automute_hook = alc283_hp_automute_hook;
                break;
        case HDA_FIXUP_ACT_INIT:
@@ -3623,9 +3660,6 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
                /* Set to manual mode */
                val = alc_read_coef_idx(codec, 0x06);
                alc_write_coef_idx(codec, 0x06, val & ~0x000c);
-               /* Enable Line1 input control by verb */
-               val = alc_read_coef_idx(codec, 0x1a);
-               alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
                break;
        }
 }
@@ -3821,6 +3855,7 @@ enum {
        ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED,
        ALC269VB_FIXUP_ORDISSIMO_EVE2,
        ALC283_FIXUP_CHROME_BOOK,
+       ALC283_FIXUP_SENSE_COMBO_JACK,
        ALC282_FIXUP_ASUS_TX300,
        ALC283_FIXUP_INT_MIC,
        ALC290_FIXUP_MONO_SPEAKERS,
@@ -4120,6 +4155,12 @@ static const struct hda_fixup alc269_fixups[] = {
                .type = HDA_FIXUP_FUNC,
                .v.func = alc283_fixup_chromebook,
        },
+       [ALC283_FIXUP_SENSE_COMBO_JACK] = {
+               .type = HDA_FIXUP_FUNC,
+               .v.func = alc283_fixup_sense_combo_jack,
+               .chained = true,
+               .chain_id = ALC283_FIXUP_CHROME_BOOK,
+       },
        [ALC282_FIXUP_ASUS_TX300] = {
                .type = HDA_FIXUP_FUNC,
                .v.func = alc282_fixup_asus_tx300,
@@ -4202,6 +4243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
        SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS),
        SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+       SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS),
        SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
        SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
        SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -4210,7 +4252,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
        SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
        SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED),
-       SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
        SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
        SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
        SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -4318,6 +4359,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
        {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
        {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
        {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
+       {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-chrome"},
+       {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
        {}
 };
 
@@ -4493,6 +4536,7 @@ enum {
        ALC861_FIXUP_AMP_VREF_0F,
        ALC861_FIXUP_NO_JACK_DETECT,
        ALC861_FIXUP_ASUS_A6RP,
+       ALC660_FIXUP_ASUS_W7J,
 };
 
 /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
@@ -4542,10 +4586,22 @@ static const struct hda_fixup alc861_fixups[] = {
                .v.func = alc861_fixup_asus_amp_vref_0f,
                .chained = true,
                .chain_id = ALC861_FIXUP_NO_JACK_DETECT,
+       },
+       [ALC660_FIXUP_ASUS_W7J] = {
+               .type = HDA_FIXUP_VERBS,
+               .v.verbs = (const struct hda_verb[]) {
+                       /* ASUS W7J needs a magic pin setup on unused NID 0x10
+                        * for enabling outputs
+                        */
+                       {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+                       { }
+               },
        }
 };
 
 static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+       SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J),
+       SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J),
        SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
        SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
        SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),

diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15afda177080da1032300bcffd18ac..1b372283bd01a947c26dc4da302f9b077adfe74a 100644 (file)
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
                goto out;
        }
 
+       snd_soc_card_set_drvdata(card, priv);
+
        card->dev = &pdev->dev;
        card->owner = THIS_MODULE;
        card->dai_link = dai;

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a618115d06171bb87314b363cc8ac9f6d..99b359e19d35f46a3c95946d976b294300c6ea70 100644 (file)
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
 
-SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
-          ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
-          ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
-          ARIZONA_OUT3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
-          ARIZONA_OUT4_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
-          ARIZONA_OUT5_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
-          ARIZONA_OUT6_OSR_SHIFT, 1, 0),
-
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
             ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
                 ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
                 0xbf, 0, digital_tlv),
 
-SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
-                      ARIZONA_OUTPUT_PATH_CONFIG_1R,
-                      ARIZONA_OUT1L_PGA_VOL_SHIFT,
-                      0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
-                      ARIZONA_OUTPUT_PATH_CONFIG_2R,
-                      ARIZONA_OUT2L_PGA_VOL_SHIFT,
-                      0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
-                      ARIZONA_OUTPUT_PATH_CONFIG_3R,
-                      ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
-
 SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
           ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759176a3bf31c5f7003a6e8cfc2d799..bc7472c968e37300edfdd38f9108bee0af77bb4f 100644 (file)
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
                iface |= 0x0001;
                break;
        case SND_SOC_DAIFMT_DSP_A:
-               iface |= 0x0003;
+               iface |= 0x0013;
                break;
        case SND_SOC_DAIFMT_DSP_B:
-               iface |= 0x0013;
+               iface |= 0x0003;
                break;
        default:
                return -EINVAL;

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 253c88bb7a4cbdb06d3d1df2862f10e80095aaa8..4f05fb88bddf1394509b9eb14c7b746909074873 100644 (file)
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
                /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
                snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+               codec->cache_sync = 1;
                break;
        }
 

diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index eb4373840bb6e1ab2cbbb1e5e9220e1c8474cb29..3665f612819d4bd2d443b0e7072cf9753ee1905b 100644 (file)
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op)
                return -ENOMEM;
 
        card->dev = &op->dev;
-       platform_set_drvdata(op, pdata);
 
        pdata->card = card;
 
@@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
        if (ret)
                dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
 
+       platform_set_drvdata(op, pdata);
+
        return ret;
 }
 

diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index d34d91743e3ffe95f9e4823bafdd33d78875b6c3..0b18f654b41340908d022003485d8a88c5e0c94e 100644 (file)
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -33,6 +33,10 @@
         SNDRV_PCM_FMTBIT_S24_LE | \
         SNDRV_PCM_FMTBIT_S32_LE)
 
+#define KIRKWOOD_SPDIF_FORMATS \
+       (SNDRV_PCM_FMTBIT_S16_LE | \
+        SNDRV_PCM_FMTBIT_S24_LE)
+
 static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
                unsigned int fmt)
 {
@@ -244,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
                                   ctl);
        }
 
-       if (dai->id == 0)
-               ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;      /* i2s */
-       else
-               ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;        /* spdif */
-
        switch (cmd) {
        case SNDRV_PCM_TRIGGER_START:
                /* configure */
                ctl = priv->ctl_play;
+               if (dai->id == 0)
+                       ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;      /* i2s */
+               else
+                       ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;        /* spdif */
+
                value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
                writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
@@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = {
                .channels_max = 2,
                .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
                                SNDRV_PCM_RATE_96000,
-               .formats = KIRKWOOD_I2S_FORMATS,
+               .formats = KIRKWOOD_SPDIF_FORMATS,
        },
        .capture = {
                .channels_min = 1,
                .channels_max = 2,
                .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
                                SNDRV_PCM_RATE_96000,
-               .formats = KIRKWOOD_I2S_FORMATS,
+               .formats = KIRKWOOD_SPDIF_FORMATS,
        },
        .ops = &kirkwood_i2s_dai_ops,
     },
@@ -493,7 +497,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
                .rates = SNDRV_PCM_RATE_8000_192000 |
                         SNDRV_PCM_RATE_CONTINUOUS |
                         SNDRV_PCM_RATE_KNOT,
-               .formats = KIRKWOOD_I2S_FORMATS,
+               .formats = KIRKWOOD_SPDIF_FORMATS,
        },
        .capture = {
                .channels_min = 1,
@@ -501,7 +505,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
                .rates = SNDRV_PCM_RATE_8000_192000 |
                         SNDRV_PCM_RATE_CONTINUOUS |
                         SNDRV_PCM_RATE_KNOT,
-               .formats = KIRKWOOD_I2S_FORMATS,
+               .formats = KIRKWOOD_SPDIF_FORMATS,
        },
        .ops = &kirkwood_i2s_dai_ops,
     },

diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6d216cb6c19bd2ba92c4d326d2ee6761dc2e5ef3..3fde9e402710f2bb16baf80b605da42772df69a0 100644 (file)
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream)
                                     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
 
        n810_ext_control(&codec->dapm);
-       return clk_enable(sys_clkout2);
+       return clk_prepare_enable(sys_clkout2);
 }
 
 static void n810_shutdown(struct snd_pcm_substream *substream)
 {
-       clk_disable(sys_clkout2);
+       clk_disable_unprepare(sys_clkout2);
 }
 
 static int n810_hw_params(struct snd_pcm_substream *substream,

diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 14011d90d70af3180f4eeedee2fa254a6d5fde25..ff60e11ecb564674bdffbe8814392fe1b2b01065 100644 (file)
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
 config SND_SOC_RCAR
        tristate "R-Car series SRU/SCU/SSIU/SSI support"
        select SND_SIMPLE_CARD
+       select REGMAP
        help
          This option enables R-Car SUR/SCU/SSIU/SSI sound support
 

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d0891a3d8437881d12d41225303cc..a66783e13a9cca182e3077b3e998c804aa40ff1d 100644 (file)
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
                        break;
                case 2:
                        ((u16 *)(&ucontrol->value.bytes.data))[0]
-                               &= ~params->mask;
+                               &= cpu_to_be16(~params->mask);
                        break;
                case 4:
                        ((u32 *)(&ucontrol->value.bytes.data))[0]
-                               &= ~params->mask;
+                               &= cpu_to_be32(~params->mask);
                        break;
                default:
                        return -EINVAL;

diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c0275fd06039efe36988b1a8ba3e8d2..3449c1e909ae10c71a695f6dc2eedecba9e8c163 100644 (file)
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res)
  */
 int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 {
-       struct device **ptr;
+       struct snd_soc_card **ptr;
        int ret;
 
        ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL);
@@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 
        ret = snd_soc_register_card(card);
        if (ret == 0) {
-               *ptr = dev;
+               *ptr = card;
                devres_add(dev, ptr);
        } else {
                devres_free(ptr);

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e41320e924efb4b93c839802f0df5722..11a90cd027faca2956172f3dfa92f6d2ee46a8fe 100644 (file)
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
        }
 }
 
-static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
+static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
        struct snd_soc_pcm_stream *codec_stream,
        struct snd_soc_pcm_stream *cpu_stream)
 {
-       hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min);
-       hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max);
+       struct snd_pcm_hardware *hw = &runtime->hw;
+
        hw->channels_min = max(codec_stream->channels_min,
                cpu_stream->channels_min);
        hw->channels_max = min(codec_stream->channels_max,
@@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
        if (cpu_stream->rates
                & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
                hw->rates |= codec_stream->rates;
+
+       snd_pcm_limit_hw_rates(runtime);
+
+       hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+       hw->rate_min = max(hw->rate_min, codec_stream->rate_min);
+       hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+       hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max);
 }
 
 /*
@@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 
        /* Check that the codec and cpu DAIs are compatible */
        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-               soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback,
+               soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback,
                        &cpu_dai_drv->playback);
        } else {
-               soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture,
+               soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture,
                        &cpu_dai_drv->capture);
        }
 
        ret = -EINVAL;
-       snd_pcm_limit_hw_rates(runtime);
        if (!runtime->hw.rates) {
                printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
                        codec_dai->name, cpu_dai->name);